refactor(audio): split monolithic audio module into submodules

Move audio engine functionality from a single large file into focused Rust modules while preserving behavior and command surface. This keeps boundaries clearer for future extraction and the upcoming lib.rs split.

src-tauri/src/audio/codec.rs

@@ -0,0 +1,21 @@
+//! Symphonia codec registry (incl. Opus) and radio decoder factory.
+use std::sync::OnceLock;
+
+use symphonia::core::codecs::{CodecRegistry, DecoderOptions};
+
+pub(crate) fn psysonic_codec_registry() -> &'static CodecRegistry {
+    static REGISTRY: OnceLock<CodecRegistry> = OnceLock::new();
+    REGISTRY.get_or_init(|| {
+        let mut registry = CodecRegistry::new();
+        symphonia::default::register_enabled_codecs(&mut registry);
+        registry.register_all::<symphonia_adapter_libopus::OpusDecoder>();
+        registry
+    })
+}
+
+pub(crate) fn try_make_radio_decoder(
+    params: &symphonia::core::codecs::CodecParameters,
+    opts: &DecoderOptions,
+) -> Result<Box<dyn symphonia::core::codecs::Decoder>, symphonia::core::errors::Error> {
+    psysonic_codec_registry().make(params, opts)
+}

src-tauri/src/audio/decode.rs

@@ -0,0 +1,682 @@
+//! Symphonia `SizedDecoder`, gapless trim, and `build_source` / `build_streaming_source`.
+use std::io::{Cursor, Read, Seek};
+use std::sync::atomic::{AtomicBool, AtomicU32, AtomicU64};
+use std::sync::Arc;
+use std::time::Duration;
+
+use rodio::source::UniformSourceIterator;
+use rodio::Source;
+use symphonia::core::{
+    audio::{AudioBufferRef, SampleBuffer, SignalSpec},
+    codecs::{DecoderOptions, CODEC_TYPE_NULL},
+    formats::{FormatOptions, FormatReader, SeekMode, SeekTo},
+    io::{MediaSource, MediaSourceStream, MediaSourceStreamOptions},
+    meta::MetadataOptions,
+    probe::Hint,
+    units::{self, Time},
+};
+
+use super::codec::{psysonic_codec_registry, try_make_radio_decoder};
+use super::sources::*;
+
+// ─── SizedCursorSource — correct byte_len for seekable in-memory sources ──────
+//
+// rodio's internal ReadSeekSource wraps Cursor<Vec<u8>> but hardcodes
+// byte_len() → None.  This tells symphonia "stream length unknown", which
+// prevents the FLAC demuxer from seeking (it validates seek offsets against
+// the total stream length from byte_len).  MP3 is unaffected because its
+// demuxer uses Xing/LAME headers instead.
+//
+// This wrapper provides the actual byte length, fixing seek for all formats.
+
+pub(crate) struct SizedCursorSource {
+    inner: Cursor<Vec<u8>>,
+    len: u64,
+}
+
+impl Read for SizedCursorSource {
+    fn read(&mut self, buf: &mut [u8]) -> std::io::Result<usize> {
+        self.inner.read(buf)
+    }
+}
+
+impl Seek for SizedCursorSource {
+    fn seek(&mut self, pos: std::io::SeekFrom) -> std::io::Result<u64> {
+        self.inner.seek(pos)
+    }
+}
+
+impl MediaSource for SizedCursorSource {
+    fn is_seekable(&self) -> bool { true }
+    fn byte_len(&self) -> Option<u64> { Some(self.len) }
+}
+
+// ─── SizedDecoder — symphonia decoder with correct byte_len ───────────────────
+//
+// Replaces rodio::Decoder::new() which wraps the source in ReadSeekSource
+// (byte_len = None).  This constructs the symphonia pipeline directly,
+// providing the correct byte_len via SizedCursorSource.
+//
+// Implements Iterator<Item = i16> + Source — identical interface to
+// rodio::Decoder, so the rest of the source chain is unchanged.
+
+/// Debug logging: codec parameters in human-readable form to verify whether
+/// playback is genuinely lossless.
+pub(crate) fn log_codec_resolution(
+    tag: &str,
+    params: &symphonia::core::codecs::CodecParameters,
+    container_hint: Option<&str>,
+) {
+    let codec_name = symphonia::default::get_codecs()
+        .get_codec(params.codec)
+        .map(|d| d.short_name)
+        .unwrap_or("?");
+    let rate = params.sample_rate.map(|r| format!("{} Hz", r)).unwrap_or_else(|| "? Hz".into());
+    let bits = params.bits_per_sample
+        .or(params.bits_per_coded_sample)
+        .map(|b| format!("{}-bit", b))
+        .unwrap_or_else(|| "?-bit".into());
+    let ch = params.channels
+        .map(|c| format!("{}ch", c.count()))
+        .unwrap_or_else(|| "?ch".into());
+    let lossless = codec_name.starts_with("pcm")
+        || matches!(
+            codec_name,
+            "flac" | "alac" | "wavpack" | "monkeys-audio" | "tta" | "shorten"
+        );
+    let kind = if lossless { "LOSSLESS" } else { "lossy" };
+    crate::app_deprintln!(
+        "[stream] {tag}: codec={codec_name} ({kind}) {bits} {rate} {ch} container={}",
+        container_hint.unwrap_or("?")
+    );
+}
+
+/// Max retries for IO/packet-read errors (fatal — network drop, truncated file).
+const DECODE_MAX_RETRIES: usize = 3;
+/// Max *consecutive* DecodeErrors before giving up on a file.
+/// Non-fatal errors like "invalid main_data offset" are silently dropped up to
+/// this limit so a handful of corrupt MP3 frames never aborts an otherwise
+/// playable track (VLC-style frame dropping).
+const MAX_CONSECUTIVE_DECODE_ERRORS: usize = 100;
+
+pub(crate) struct SizedDecoder {
+    decoder: Box<dyn symphonia::core::codecs::Decoder>,
+    current_frame_offset: usize,
+    format: Box<dyn FormatReader>,
+    total_duration: Option<Time>,
+    buffer: SampleBuffer<i16>,
+    spec: SignalSpec,
+    /// Counts consecutive DecodeErrors in the hot-path. Reset to 0 on every
+    /// successfully decoded frame. Used to detect fully undecodable streams.
+    consecutive_decode_errors: usize,
+}
+
+impl SizedDecoder {
+    pub(crate) fn new(data: Vec<u8>, format_hint: Option<&str>, hi_res: bool) -> Result<Self, String> {
+        let data_len = data.len() as u64;
+        let source = SizedCursorSource {
+            inner: Cursor::new(data),
+            len: data_len,
+        };
+        // Hi-Res: 4 MB read-ahead so Symphonia demuxes fewer Read calls for
+        // high-bitrate files (88.2 kHz/24-bit FLAC ≈ 1800 kbps).
+        // Standard: 512 KB is plenty for MP3/AAC — larger buffers waste allocation
+        // and compete with the playback thread at track start.
+        let buf_len = if hi_res { 4 * 1024 * 1024 } else { 512 * 1024 };
+        let mss = MediaSourceStream::new(
+            Box::new(source) as Box<dyn MediaSource>,
+            MediaSourceStreamOptions { buffer_len: buf_len },
+        );
+
+        let mut hint = Hint::new();
+        if let Some(ext) = format_hint {
+            hint.with_extension(ext);
+        }
+        let format_opts = FormatOptions {
+            // Disable gapless parsing — Symphonia 0.5.5 crashes on `edts` atoms
+            // present in older iTunes-purchased M4A files.
+            enable_gapless: false,
+            ..Default::default()
+        };
+
+        let meta_opts = symphonia::core::meta::MetadataOptions {
+            // Cap embedded cover art at 8 MiB so oversized MJPEG images in
+            // iTunes M4A files don't choke the parser.
+            limit_visual_bytes: symphonia::core::meta::Limit::Maximum(8 * 1024 * 1024),
+            ..Default::default()
+        };
+
+        let probed = symphonia::default::get_probe()
+            .format(&hint, mss, &format_opts, &meta_opts)
+            .map_err(|e| {
+                let hint_str = format_hint.unwrap_or("unknown");
+                // Always print the raw Symphonia error to the terminal for diagnosis.
+                crate::app_eprintln!("[psysonic] probe failed (hint={hint_str}): {e}");
+                if e.to_string().to_lowercase().contains("unsupported") {
+                    format!("unsupported format: .{hint_str} files cannot be played (no demuxer)")
+                } else {
+                    format!("could not open audio stream (.{hint_str}): {e}")
+                }
+            })?;
+
+        let track = probed.format
+            .tracks()
+            .iter()
+            // Explicitly select only audio tracks: must have a valid codec and a
+            // sample_rate. This skips MJPEG cover-art streams that iTunes M4A
+            // files embed as a secondary video track.
+            .find(|t| {
+                t.codec_params.codec != CODEC_TYPE_NULL
+                    && t.codec_params.sample_rate.is_some()
+            })
+            .ok_or_else(|| {
+                crate::app_eprintln!("[psysonic] no audio track found among {} tracks", probed.format.tracks().len());
+                "no playable audio track found in file".to_string()
+            })?;
+
+        let track_id = track.id;
+        let total_duration = track.codec_params.time_base
+            .zip(track.codec_params.n_frames)
+            .map(|(base, frames)| base.calc_time(frames));
+
+        log_codec_resolution("bytes", &track.codec_params, format_hint);
+
+        let mut decoder = psysonic_codec_registry()
+            .make(&track.codec_params, &DecoderOptions::default())
+            .map_err(|e| {
+                crate::app_eprintln!("[psysonic] codec init failed: {e}");
+                if e.to_string().to_lowercase().contains("unsupported") {
+                    "unsupported codec: no decoder available for this audio format".to_string()
+                } else {
+                    format!("failed to initialise audio decoder: {e}")
+                }
+            })?;
+
+        let mut format = probed.format;
+
+        // Decode the first packet to initialise spec + buffer.
+        // DecodeErrors (e.g. "invalid main_data offset") are non-fatal: drop the
+        // frame and try the next packet up to MAX_CONSECUTIVE_DECODE_ERRORS times.
+        let mut decode_errors: usize = 0;
+        let decoded = loop {
+            let packet = match format.next_packet() {
+                Ok(p) => p,
+                Err(symphonia::core::errors::Error::IoError(_)) => {
+                    break decoder.last_decoded();
+                }
+                Err(e) => {
+                    crate::app_eprintln!("[psysonic] next_packet error: {e}");
+                    return Err(format!("could not read audio data: {e}"));
+                }
+            };
+            if packet.track_id() != track_id {
+                crate::app_eprintln!("[psysonic] skipping packet for track {} (want {})", packet.track_id(), track_id);
+                continue;
+            }
+            match decoder.decode(&packet) {
+                Ok(decoded) => break decoded,
+                Err(symphonia::core::errors::Error::DecodeError(ref msg)) => {
+                    decode_errors += 1;
+                    crate::app_eprintln!("[psysonic] init: dropped corrupt frame #{decode_errors}: {msg}");
+                    if decode_errors >= MAX_CONSECUTIVE_DECODE_ERRORS {
+                        return Err("too many consecutive decode errors during init — file may be corrupt".into());
+                    }
+                }
+                Err(e) => {
+                    crate::app_eprintln!("[psysonic] fatal decode error: {e}");
+                    return Err(format!("audio decode error: {e}"));
+                }
+            }
+        };
+
+        let spec = decoded.spec().to_owned();
+        let buffer = Self::make_buffer(decoded, &spec);
+
+        Ok(SizedDecoder {
+            decoder,
+            current_frame_offset: 0,
+            format,
+            total_duration,
+            buffer,
+            spec,
+            consecutive_decode_errors: 0,
+        })
+    }
+
+    /// Build a decoder from any `MediaSource` (e.g. track-stream or radio).
+    /// Uses `enable_gapless: false` — live streams are not seekable; gapless
+    /// trimming requires seeking to read the LAME/iTunSMPB end-padding info.
+    pub(crate) fn new_streaming(
+        media: Box<dyn MediaSource>,
+        format_hint: Option<&str>,
+        source_tag: &str,
+    ) -> Result<Self, String> {
+        // Larger read-ahead buffer for the live streaming SPSC consumer — reduces
+        // read() call frequency into the ring buffer, easing I/O spikes.
+        let mss = MediaSourceStream::new(media, MediaSourceStreamOptions { buffer_len: 512 * 1024 });
+        let mut hint = Hint::new();
+        if let Some(ext) = format_hint { hint.with_extension(ext); }
+        let format_opts = FormatOptions { enable_gapless: false, ..Default::default() };
+        let probed = symphonia::default::get_probe()
+            .format(&hint, mss, &format_opts, &MetadataOptions::default())
+            .map_err(|e| format!("{source_tag}: format probe failed: {e}"))?;
+
+        let track = probed.format.tracks().iter()
+            .find(|t| t.codec_params.codec != CODEC_TYPE_NULL)
+            .ok_or_else(|| format!("{source_tag}: no audio track found"))?;
+        let track_id = track.id;
+        log_codec_resolution(source_tag, &track.codec_params, format_hint);
+        // Live streams have no known total frame count → total_duration = None.
+        let total_duration = None;
+        let mut decoder = try_make_radio_decoder(&track.codec_params, &DecoderOptions::default())
+            .map_err(|e| format!("{source_tag}: codec init failed: {e}"))?;
+        let mut format = probed.format;
+
+        let mut errors = 0usize;
+        let decoded = loop {
+            let packet = match format.next_packet() {
+                Ok(p) => p,
+                Err(_) => break decoder.last_decoded(),
+            };
+            if packet.track_id() != track_id { continue; }
+            match decoder.decode(&packet) {
+                Ok(d) => break d,
+                Err(symphonia::core::errors::Error::DecodeError(ref msg)) => {
+                    errors += 1;
+                    crate::app_eprintln!("[psysonic] {source_tag} init: dropped corrupt frame #{errors}: {msg}");
+                    if errors >= MAX_CONSECUTIVE_DECODE_ERRORS {
+                        return Err(format!("{source_tag}: too many consecutive decode errors"));
+                    }
+                }
+                Err(e) => return Err(format!("{source_tag}: decode error: {e}")),
+            }
+        };
+        let spec = decoded.spec().to_owned();
+        let buffer = Self::make_buffer(decoded, &spec);
+        Ok(SizedDecoder { decoder, current_frame_offset: 0, format, total_duration, buffer, spec, consecutive_decode_errors: 0 })
+    }
+
+    #[inline]
+    fn make_buffer(decoded: AudioBufferRef, spec: &SignalSpec) -> SampleBuffer<i16> {
+        let duration = units::Duration::from(decoded.capacity() as u64);
+        let mut buffer = SampleBuffer::<i16>::new(duration, *spec);
+        buffer.copy_interleaved_ref(decoded);
+        buffer
+    }
+
+    /// Refine position after a coarse seek — decode packets until we reach the
+    /// exact requested timestamp.
+    fn refine_position(
+        &mut self,
+        seek_res: symphonia::core::formats::SeekedTo,
+    ) -> Result<(), String> {
+        let mut samples_to_pass = seek_res.required_ts - seek_res.actual_ts;
+        let packet = loop {
+            let candidate = self.format.next_packet()
+                .map_err(|e| format!("refine seek: {e}"))?;
+            if candidate.dur() > samples_to_pass {
+                break candidate;
+            }
+            samples_to_pass -= candidate.dur();
+        };
+
+        let mut decoded = self.decoder.decode(&packet);
+        for _ in 0..DECODE_MAX_RETRIES {
+            if decoded.is_err() {
+                let p = self.format.next_packet()
+                    .map_err(|e| format!("refine retry: {e}"))?;
+                decoded = self.decoder.decode(&p);
+            }
+        }
+
+        let decoded = decoded.map_err(|e| format!("refine decode: {e}"))?;
+        decoded.spec().clone_into(&mut self.spec);
+        self.buffer = Self::make_buffer(decoded, &self.spec);
+        self.current_frame_offset = samples_to_pass as usize * self.spec.channels.count();
+        Ok(())
+    }
+}
+
+impl Iterator for SizedDecoder {
+    type Item = i16;
+
+    #[inline]
+    fn next(&mut self) -> Option<i16> {
+        if self.current_frame_offset >= self.buffer.len() {
+            // Loop until a decodable packet is found or the stream ends.
+            // DecodeErrors (e.g. MP3 "invalid main_data offset") are non-fatal:
+            // drop the frame and advance to the next packet. IO errors and a
+            // clean end-of-stream both terminate the iterator normally.
+            loop {
+                let packet = self.format.next_packet().ok()?;
+                match self.decoder.decode(&packet) {
+                    Ok(decoded) => {
+                        self.consecutive_decode_errors = 0;
+                        decoded.spec().clone_into(&mut self.spec);
+                        self.buffer = Self::make_buffer(decoded, &self.spec);
+                        self.current_frame_offset = 0;
+                        break;
+                    }
+                    Err(symphonia::core::errors::Error::DecodeError(ref msg)) => {
+                        #[cfg(not(debug_assertions))]
+                        let _ = msg;
+                        self.consecutive_decode_errors += 1;
+                        // Log sparingly: first drop, then every 10th to avoid spam.
+                        if self.consecutive_decode_errors == 1
+                            || self.consecutive_decode_errors % 10 == 0
+                        {
+                            crate::app_deprintln!(
+                                "[psysonic] dropped corrupt frame #{}: {msg}",
+                                self.consecutive_decode_errors
+                            );
+                        }
+                        if self.consecutive_decode_errors >= MAX_CONSECUTIVE_DECODE_ERRORS {
+                            crate::app_deprintln!(
+                                "[psysonic] {MAX_CONSECUTIVE_DECODE_ERRORS} consecutive decode \
+                                 failures — stream appears unrecoverable, stopping"
+                            );
+                            return None;
+                        }
+                        // continue → fetch next packet
+                    }
+                    Err(_) => return None, // IO error or fatal codec error → end of stream
+                }
+            }
+        }
+
+        let sample = *self.buffer.samples().get(self.current_frame_offset)?;
+        self.current_frame_offset += 1;
+        Some(sample)
+    }
+}
+
+impl Source for SizedDecoder {
+    #[inline]
+    fn current_frame_len(&self) -> Option<usize> {
+        Some(self.buffer.samples().len())
+    }
+
+    #[inline]
+    fn channels(&self) -> u16 {
+        self.spec.channels.count() as u16
+    }
+
+    #[inline]
+    fn sample_rate(&self) -> u32 {
+        self.spec.rate
+    }
+
+    #[inline]
+    fn total_duration(&self) -> Option<Duration> {
+        self.total_duration.map(|Time { seconds, frac }| {
+            Duration::new(seconds, (frac * 1_000_000_000.0) as u32)
+        })
+    }
+
+    fn try_seek(&mut self, pos: Duration) -> Result<(), rodio::source::SeekError> {
+        let seek_beyond_end = self
+            .total_duration()
+            .is_some_and(|dur| dur.saturating_sub(pos).as_millis() < 1);
+
+        let time: Time = if seek_beyond_end {
+            let t = self.total_duration.unwrap_or(pos.as_secs_f64().into());
+            // Step back a tiny bit — some demuxers can't seek to the exact end.
+            let mut secs = t.seconds;
+            let mut frac = t.frac - 0.0001;
+            if frac < 0.0 {
+                secs = secs.saturating_sub(1);
+                frac = 1.0 - frac;
+            }
+            Time { seconds: secs, frac }
+        } else {
+            pos.as_secs_f64().into()
+        };
+
+        let to_skip = self.current_frame_offset % self.channels() as usize;
+
+        let seek_res = self
+            .format
+            .seek(SeekMode::Accurate, SeekTo::Time { time, track_id: None })
+            .map_err(|e| rodio::source::SeekError::Other(
+                Box::new(std::io::Error::new(std::io::ErrorKind::Other, e.to_string()))
+            ))?;
+
+        self.refine_position(seek_res)
+            .map_err(|e| rodio::source::SeekError::Other(
+                Box::new(std::io::Error::new(std::io::ErrorKind::Other, e))
+            ))?;
+
+        self.current_frame_offset += to_skip;
+        Ok(())
+    }
+}
+
+// ─── Encoder-gap trimming (iTunSMPB) ─────────────────────────────────────────
+//
+// MP3/AAC encoders prepend an "encoder delay" (typically 576–2112 silent
+// samples for LAME) and append end-padding to fill the final frame.
+// iTunes embeds the exact counts in an ID3v2 COMM frame with description
+// "iTunSMPB". Format: " 00000000 DELAY PADDING TOTAL ..."  (space-separated hex)
+//
+// Parsing strategy: scan raw bytes for the ASCII marker, then extract the
+// first whitespace-separated hex tokens after it.
+
+pub(crate) struct GaplessInfo {
+    delay_samples: u64,
+    total_valid_samples: Option<u64>,
+}
+
+impl Default for GaplessInfo {
+    fn default() -> Self {
+        Self { delay_samples: 0, total_valid_samples: None }
+    }
+}
+
+pub(crate) fn find_subsequence(data: &[u8], needle: &[u8]) -> Option<usize> {
+    data.windows(needle.len()).position(|w| w == needle)
+}
+
+pub(crate) fn parse_gapless_info(data: &[u8]) -> GaplessInfo {
+    let pos = match find_subsequence(data, b"iTunSMPB") {
+        Some(p) => p,
+        None => return GaplessInfo::default(),
+    };
+
+    // In M4A/iTunes files the key is followed by a binary 'data' atom header
+    // (16 bytes: size[4] + "data"[4] + type_flags[4] + locale[4]) before the
+    // actual value string. Search for the " 00000000 " sentinel that every
+    // iTunSMPB value starts with to locate the true start of the text.
+    let search_end = data.len().min(pos + 8 + 128);
+    let search_window = &data[pos + 8..search_end];
+    let value_start = find_subsequence(search_window, b" 00000000 ")
+        .map(|off| pos + 8 + off)
+        .unwrap_or(pos + 8);
+
+    let tail = &data[value_start..data.len().min(value_start + 256)];
+    let text: String = tail.iter()
+        .map(|&b| b as char)
+        .filter(|c| c.is_ascii_hexdigit() || *c == ' ')
+        .collect();
+
+    let parts: Vec<&str> = text.split_whitespace().collect();
+    // parts[0] = "00000000", parts[1] = delay, parts[2] = padding, parts[3] = total
+    if parts.len() < 3 {
+        return GaplessInfo::default();
+    }
+    let delay = u64::from_str_radix(parts.get(1).unwrap_or(&"0"), 16).unwrap_or(0);
+    let padding = u64::from_str_radix(parts.get(2).unwrap_or(&"0"), 16).unwrap_or(0);
+    let total_raw = parts.get(3).and_then(|s| u64::from_str_radix(s, 16).ok());
+
+    let total_valid = total_raw.map(|t| t).filter(|&t| t > 0).or_else(|| {
+        // Derive from delay + padding if total not available:
+        // Not possible without knowing total encoded samples, so just use None.
+        let _ = padding;
+        None
+    });
+
+    GaplessInfo { delay_samples: delay, total_valid_samples: total_valid }
+}
+
+/// Result of build_source: the fully-wrapped source plus metadata and control Arcs.
+pub(crate) struct BuiltSource {
+    pub(crate) source: CountingSource<NotifyingSource<TriggeredFadeOut<EqualPowerFadeIn<EqSource<DynSource>>>>>,
+    pub(crate) duration_secs: f64,
+    pub(crate) output_rate: u32,
+    pub(crate) output_channels: u16,
+    /// Trigger for the sample-level crossfade fade-out.
+    pub(crate) fadeout_trigger: Arc<AtomicBool>,
+    /// Total samples for the fade-out (set before triggering).
+    pub(crate) fadeout_samples: Arc<AtomicU64>,
+}
+
+/// Build a fully-prepared playback source:
+///   decode → trim → resample → EQ → fade-in → triggered-fade-out → notify → count
+///
+/// `fade_in_dur`:
+///   • `Duration::ZERO`          — unity gain; used for gapless chain (no click)
+///   • `Duration::from_millis(5)` — micro-fade; used for hard cuts (anti-click)
+///   • `Duration::from_secs_f32(cf)` — full equal-power fade-in for crossfade
+///
+/// `sample_counter`: atomic counter incremented per sample for drift-free position.
+/// `target_rate`: canonical output sample rate for resampling (0 = no resampling).
+/// `format_hint`: optional file extension (e.g. "flac", "mp3") to help symphonia probe.
+pub(crate) fn build_source(
+    data: Vec<u8>,
+    duration_hint: f64,
+    eq_gains: Arc<[AtomicU32; 10]>,
+    eq_enabled: Arc<AtomicBool>,
+    eq_pre_gain: Arc<AtomicU32>,
+    done_flag: Arc<AtomicBool>,
+    fade_in_dur: Duration,
+    sample_counter: Arc<AtomicU64>,
+    target_rate: u32,
+    format_hint: Option<&str>,
+    hi_res: bool,
+) -> Result<BuiltSource, String> {
+    let gapless = parse_gapless_info(&data);
+
+    let decoder = SizedDecoder::new(data, format_hint, hi_res)?;
+    let sample_rate = decoder.sample_rate();
+    let channels = decoder.channels();
+
+    // Determine effective duration.
+    // Prefer hint from Subsonic API (reliable) over decoder (unreliable for VBR MP3).
+    let effective_dur = if duration_hint > 1.0 {
+        duration_hint
+    } else {
+        decoder.total_duration()
+            .map(|d| d.as_secs_f64())
+            .unwrap_or(duration_hint)
+    };
+
+    // Apply encoder-delay trim and optional end-padding trim,
+    // then resample to the canonical target rate if needed.
+    let dyn_src: DynSource = if gapless.delay_samples > 0 || gapless.total_valid_samples.is_some() {
+        let delay_dur = Duration::from_secs_f64(
+            gapless.delay_samples as f64 / sample_rate as f64
+        );
+        let base = decoder.convert_samples::<f32>().skip_duration(delay_dur);
+
+        if let Some(total) = gapless.total_valid_samples {
+            let valid_dur = Duration::from_secs_f64(total as f64 / sample_rate as f64);
+            let trimmed = base.take_duration(valid_dur);
+            if target_rate > 0 && sample_rate != target_rate {
+                DynSource::new(UniformSourceIterator::new(trimmed, channels, target_rate))
+            } else {
+                DynSource::new(trimmed)
+            }
+        } else {
+            if target_rate > 0 && sample_rate != target_rate {
+                DynSource::new(UniformSourceIterator::new(base, channels, target_rate))
+            } else {
+                DynSource::new(base)
+            }
+        }
+    } else {
+        let converted = decoder.convert_samples::<f32>();
+        if target_rate > 0 && sample_rate != target_rate {
+            DynSource::new(UniformSourceIterator::new(converted, channels, target_rate))
+        } else {
+            DynSource::new(converted)
+        }
+    };
+
+    let output_rate = if target_rate > 0 && sample_rate != target_rate { target_rate } else { sample_rate };
+
+    let fadeout_trigger = Arc::new(AtomicBool::new(false));
+    let fadeout_samples = Arc::new(AtomicU64::new(0));
+
+    let eq_src = EqSource::new(dyn_src, eq_gains, eq_enabled, eq_pre_gain);
+    let fade_in = EqualPowerFadeIn::new(eq_src, fade_in_dur);
+    let fade_out = TriggeredFadeOut::new(fade_in, fadeout_trigger.clone(), fadeout_samples.clone());
+    let notifying = NotifyingSource::new(fade_out, done_flag);
+    let counting = CountingSource::new(notifying, sample_counter);
+
+    Ok(BuiltSource {
+        source: counting,
+        duration_secs: effective_dur,
+        output_rate,
+        output_channels: channels,
+        fadeout_trigger,
+        fadeout_samples,
+    })
+}
+
+/// Streaming variant of `build_source`: uses a live `SizedDecoder` source
+/// (non-seekable) and skips iTunSMPB parsing, but preserves the same EQ/fade/
+/// counting wrappers and output metadata.
+pub(crate) fn build_streaming_source(
+    decoder: SizedDecoder,
+    duration_hint: f64,
+    eq_gains: Arc<[AtomicU32; 10]>,
+    eq_enabled: Arc<AtomicBool>,
+    eq_pre_gain: Arc<AtomicU32>,
+    done_flag: Arc<AtomicBool>,
+    fade_in_dur: Duration,
+    sample_counter: Arc<AtomicU64>,
+    target_rate: u32,
+) -> Result<BuiltSource, String> {
+    let sample_rate = decoder.sample_rate();
+    let channels = decoder.channels();
+
+    // For streaming starts prefer server-provided duration when available.
+    let effective_dur = if duration_hint > 1.0 {
+        duration_hint
+    } else {
+        decoder
+            .total_duration()
+            .map(|d| d.as_secs_f64())
+            .unwrap_or(duration_hint)
+    };
+
+    let converted = decoder.convert_samples::<f32>();
+    let dyn_src: DynSource = if target_rate > 0 && sample_rate != target_rate {
+        DynSource::new(UniformSourceIterator::new(converted, channels, target_rate))
+    } else {
+        DynSource::new(converted)
+    };
+
+    let output_rate = if target_rate > 0 && sample_rate != target_rate {
+        target_rate
+    } else {
+        sample_rate
+    };
+
+    let fadeout_trigger = Arc::new(AtomicBool::new(false));
+    let fadeout_samples = Arc::new(AtomicU64::new(0));
+
+    let eq_src = EqSource::new(dyn_src, eq_gains, eq_enabled, eq_pre_gain);
+    let fade_in = EqualPowerFadeIn::new(eq_src, fade_in_dur);
+    let fade_out = TriggeredFadeOut::new(fade_in, fadeout_trigger.clone(), fadeout_samples.clone());
+    let notifying = NotifyingSource::new(fade_out, done_flag);
+    let counting = CountingSource::new(notifying, sample_counter);
+
+    Ok(BuiltSource {
+        source: counting,
+        duration_secs: effective_dur,
+        output_rate,
+        output_channels: channels,
+        fadeout_trigger,
+        fadeout_samples,
+    })
+}

src-tauri/src/audio/dev_io.rs

@@ -0,0 +1,88 @@
+//! Output device enumeration with suppressed ALSA stderr noise.
+#[cfg(unix)]
+use libc;
+// `rodio::cpal` is referenced from the included body.
+
+/// ALSA probes noisy plugins during device queries — suppress stderr on Unix.
+#[cfg(unix)]
+pub(crate) fn with_suppressed_alsa_stderr<R>(f: impl FnOnce() -> R) -> R {
+    struct StderrGuard(i32);
+    impl Drop for StderrGuard {
+        fn drop(&mut self) {
+            unsafe { libc::dup2(self.0, 2); libc::close(self.0); }
+        }
+    }
+    let _guard = unsafe {
+        let saved = libc::dup(2);
+        let devnull = libc::open(b"/dev/null\0".as_ptr() as *const libc::c_char, libc::O_WRONLY);
+        libc::dup2(devnull, 2);
+        libc::close(devnull);
+        StderrGuard(saved)
+    };
+    f()
+}
+
+#[cfg(not(unix))]
+#[inline]
+pub(crate) fn with_suppressed_alsa_stderr<R>(f: impl FnOnce() -> R) -> R {
+    f()
+}
+
+pub(crate) fn enumerate_output_device_names() -> Vec<String> {
+    use rodio::cpal::traits::{DeviceTrait, HostTrait};
+    with_suppressed_alsa_stderr(|| {
+        let host = rodio::cpal::default_host();
+        host.output_devices()
+            .map(|iter| iter.filter_map(|d| d.name().ok()).collect())
+            .unwrap_or_default()
+    })
+}
+
+/// Linux ALSA-style cpal names: same physical sink can appear with different suffixes;
+/// busy devices are sometimes omitted from `output_devices()` while playback works.
+#[cfg(target_os = "linux")]
+pub(crate) fn linux_alsa_sink_fingerprint(name: &str) -> Option<(String, String, u32)> {
+    const IFACES: &[&str] = &[
+        "hdmi", "hw", "plughw", "sysdefault", "iec958", "front", "dmix", "surround40",
+        "surround51", "surround71",
+    ];
+    let colon = name.find(':')?;
+    let iface = name[..colon].to_ascii_lowercase();
+    if !IFACES.iter().any(|&i| i == iface.as_str()) {
+        return None;
+    }
+    let card = name.split("CARD=").nth(1)?.split(',').next()?.to_string();
+    let dev = name
+        .split("DEV=")
+        .nth(1)
+        .and_then(|s| s.split(',').next())
+        .and_then(|s| s.parse().ok())
+        .unwrap_or(0);
+    Some((iface, card, dev))
+}
+
+#[cfg(not(target_os = "linux"))]
+#[inline]
+pub(crate) fn linux_alsa_sink_fingerprint(_name: &str) -> Option<(String, String, u32)> {
+    None
+}
+
+pub(crate) fn output_devices_logically_same(a: &str, b: &str) -> bool {
+    if a == b {
+        return true;
+    }
+    match (
+        linux_alsa_sink_fingerprint(a),
+        linux_alsa_sink_fingerprint(b),
+    ) {
+        (Some(fa), Some(fb)) => fa == fb,
+        _ => false,
+    }
+}
+
+/// True if `pinned` is the same sink as some entry (exact or Linux ALSA logical match).
+pub(crate) fn output_enumeration_includes_pinned(available: &[String], pinned: &str) -> bool {
+    available
+        .iter()
+        .any(|d| output_devices_logically_same(d, pinned))
+}

src-tauri/src/audio/device_watcher.rs

@@ -0,0 +1,150 @@
+//! Poll default output device and pinned-device presence; reopen stream when needed.
+use std::sync::atomic::Ordering;
+use std::time::Duration;
+
+use tauri::Emitter;
+
+use super::engine::AudioEngine;
+
+pub fn start_device_watcher(engine: &AudioEngine, app: tauri::AppHandle) {
+    let reopen_tx       = engine.stream_reopen_tx.clone();
+    let stream_handle   = engine.stream_handle.clone();
+    let stream_rate     = engine.stream_sample_rate.clone();
+    let current         = engine.current.clone();
+    let fading_out      = engine.fading_out_sink.clone();
+    let selected_device = engine.selected_device.clone();
+
+    tauri::async_runtime::spawn(async move {
+        let mut last_default: Option<String> = tauri::async_runtime::spawn_blocking(|| {
+            use rodio::cpal::traits::{DeviceTrait, HostTrait};
+            rodio::cpal::default_host()
+                .default_output_device()
+                .and_then(|d| d.name().ok())
+        }).await.unwrap_or(None);
+
+        // macOS/Windows: consecutive polls where a pinned device is absent from cpal's list.
+        #[cfg(not(target_os = "linux"))]
+        let mut pinned_miss_count: u32 = 0;
+
+        loop {
+            tokio::time::sleep(Duration::from_secs(3)).await;
+
+            // Enumerate all available output devices and the current default.
+            // Suppress stderr on Unix to avoid ALSA probing noise (JACK, OSS, dmix).
+            let (current_default, available) = tauri::async_runtime::spawn_blocking(|| {
+                use rodio::cpal::traits::{DeviceTrait, HostTrait};
+                #[cfg(unix)]
+                let _guard = unsafe {
+                    struct StderrGuard(i32);
+                    impl Drop for StderrGuard {
+                        fn drop(&mut self) { unsafe { libc::dup2(self.0, 2); libc::close(self.0); } }
+                    }
+                    let saved = libc::dup(2);
+                    let devnull = libc::open(b"/dev/null\0".as_ptr() as *const libc::c_char, libc::O_WRONLY);
+                    libc::dup2(devnull, 2);
+                    libc::close(devnull);
+                    StderrGuard(saved)
+                };
+                let host = rodio::cpal::default_host();
+                let default = host.default_output_device().and_then(|d| d.name().ok());
+                let available: Vec<String> = host
+                    .output_devices()
+                    .map(|iter| iter.filter_map(|d| d.name().ok()).collect())
+                    .unwrap_or_default();
+                (default, available)
+            }).await.unwrap_or((None, vec![]));
+
+            // Empty list almost always means a transient enumeration failure, not
+            // that every output device vanished. Treating it as "pinned missing"
+            // caused false audio:device-reset (UI jumped back to system default)
+            // when switching to external USB / class-compliant interfaces.
+            if available.is_empty() {
+                continue;
+            }
+
+            let pinned = selected_device.lock().unwrap().clone();
+
+            #[cfg(target_os = "linux")]
+            if pinned.is_some() {
+                // Do not infer "unplugged" from `output_devices()` when a device is pinned.
+                // ALSA/cpal often omit the active HDMI/USB sink from enumeration for the
+                // whole session — any miss counter eventually tripped audio:device-reset.
+                // Clearing the pin is left to the user (Settings → System Default) or
+                // to a future explicit error signal from the output stream.
+                continue;
+            }
+
+            // ── Case 2 (non-Linux): pinned device disappeared from enumeration ─
+            #[cfg(not(target_os = "linux"))]
+            if let Some(ref dev_name) = pinned {
+                if !output_enumeration_includes_pinned(&available, dev_name) {
+                    pinned_miss_count += 1;
+                    if pinned_miss_count < 3 {
+                        continue;
+                    }
+                    crate::app_eprintln!("[psysonic] device-watcher: pinned device '{dev_name}' disconnected, falling back to system default");
+                    pinned_miss_count = 0;
+                    *selected_device.lock().unwrap() = None;
+
+                    tokio::time::sleep(Duration::from_millis(500)).await;
+
+                    let rate = stream_rate.load(Ordering::Relaxed);
+                    let reopen_tx2 = reopen_tx.clone();
+                    let new_handle = tauri::async_runtime::spawn_blocking(move || {
+                        let (reply_tx, reply_rx) =
+                            std::sync::mpsc::sync_channel::<rodio::OutputStreamHandle>(0);
+                        if reopen_tx2.send((rate, false, None, reply_tx)).is_err() {
+                            return None;
+                        }
+                        reply_rx.recv_timeout(Duration::from_secs(5)).ok()
+                    }).await.unwrap_or(None);
+
+                    if let Some(handle) = new_handle {
+                        *stream_handle.lock().unwrap() = handle;
+                        if let Some(s) = current.lock().unwrap().sink.take() { s.stop(); }
+                        if let Some(s) = fading_out.lock().unwrap().take()   { s.stop(); }
+                        app.emit("audio:device-reset", ()).ok();
+                    }
+
+                    last_default = current_default;
+                } else {
+                    pinned_miss_count = 0;
+                }
+                continue;
+            }
+
+            // ── Case 1: no pinned device, system default changed ──────────────
+            if current_default == last_default {
+                continue;
+            }
+
+            last_default = current_default.clone();
+
+            let Some(_new_name) = current_default else { continue };
+
+            // Debounce: give the OS time to finish configuring the new device.
+            tokio::time::sleep(Duration::from_millis(500)).await;
+
+            let rate = stream_rate.load(Ordering::Relaxed);
+            let reopen_tx2 = reopen_tx.clone();
+            let new_handle = tauri::async_runtime::spawn_blocking(move || {
+                let (reply_tx, reply_rx) =
+                    std::sync::mpsc::sync_channel::<rodio::OutputStreamHandle>(0);
+                if reopen_tx2.send((rate, false, None, reply_tx)).is_err() {
+                    return None;
+                }
+                reply_rx.recv_timeout(Duration::from_secs(5)).ok()
+            }).await.unwrap_or(None);
+
+            let Some(handle) = new_handle else {
+                crate::app_eprintln!("[psysonic] device-watcher: stream reopen timed out");
+                continue;
+            };
+
+            *stream_handle.lock().unwrap() = handle;
+            if let Some(s) = current.lock().unwrap().sink.take() { s.stop(); }
+            if let Some(s) = fading_out.lock().unwrap().take()   { s.stop(); }
+            app.emit("audio:device-changed", ()).ok();
+        }
+    });
+}

src-tauri/src/audio/engine.rs

@@ -0,0 +1,429 @@
+//! `AudioEngine` / `AudioCurrent`, stream thread, and HTTP client refresh.
+#[cfg(unix)]
+use libc;
+use std::sync::atomic::{AtomicBool, AtomicU32, AtomicU64};
+use std::sync::{Arc, Mutex, RwLock};
+use std::time::{Duration, Instant};
+
+use rodio::Sink;
+use tauri::{AppHandle, Manager};
+
+use super::state::{ChainedInfo, PreloadedTrack};
+
+pub struct AudioEngine {
+    pub stream_handle: Arc<std::sync::Mutex<rodio::OutputStreamHandle>>,
+    /// Sample rate the output stream was last opened at (updated on every re-open).
+    pub stream_sample_rate: Arc<AtomicU32>,
+    /// The rate the device was opened at on cold start — used to restore the
+    /// stream when Hi-Res is toggled off while a hi-res rate is active.
+    pub device_default_rate: u32,
+    /// Sends `(desired_rate, is_hi_res, device_name, reply_tx)` to the audio-stream
+    /// thread to re-open the output device. `device_name = None` → system default.
+    pub stream_reopen_tx: std::sync::mpsc::SyncSender<(u32, bool, Option<String>, std::sync::mpsc::SyncSender<rodio::OutputStreamHandle>)>,
+    /// User-selected output device name (None = follow system default).
+    pub selected_device: Arc<Mutex<Option<String>>>,
+    pub current: Arc<Mutex<AudioCurrent>>,
+    /// Monotonically incremented on each audio_play (non-chain) / audio_stop call.
+    pub generation: Arc<AtomicU64>,
+    pub http_client: Arc<RwLock<reqwest::Client>>,
+    pub eq_gains: Arc<[AtomicU32; 10]>,
+    pub eq_enabled: Arc<AtomicBool>,
+    pub eq_pre_gain: Arc<AtomicU32>,
+    pub(crate) preloaded: Arc<Mutex<Option<PreloadedTrack>>>,
+    /// Last fully downloaded manual-stream track bytes (same playback identity),
+    /// used to recover seek/replay without waiting for network again.
+    pub(crate) stream_completed_cache: Arc<Mutex<Option<PreloadedTrack>>>,
+    /// True when the currently playing source supports seeking (in-memory bytes
+    /// or `RangedHttpSource`); false for the legacy non-seekable streaming
+    /// fallback (`AudioStreamReader`). `audio_seek` rejects with a "not
+    /// seekable" error when false so the frontend restart-fallback can engage.
+    pub(crate) current_is_seekable: Arc<AtomicBool>,
+    pub crossfade_enabled: Arc<AtomicBool>,
+    pub crossfade_secs: Arc<AtomicU32>,
+    pub fading_out_sink: Arc<Mutex<Option<Arc<Sink>>>>,
+    /// When true, audio_play chains sources to the existing Sink instead of
+    /// creating a new one, achieving sample-accurate gapless transitions.
+    pub gapless_enabled: Arc<AtomicBool>,
+    /// 0=off, 1=replaygain, 2=loudness (future runtime loudness engine).
+    pub normalization_engine: Arc<AtomicU32>,
+    /// Target loudness in LUFS for loudness engine (future use).
+    pub normalization_target_lufs: Arc<AtomicU32>,
+    /// Extra attenuation (dB) when no loudness DB row exists at decode bind; also seeds streaming heuristics (Settings).
+    pub loudness_pre_analysis_attenuation_db: Arc<AtomicU32>,
+    /// Info about the next-up chained track (gapless mode).
+    /// The progress task reads this when `current_source_done` fires.
+    pub(crate) chained_info: Arc<Mutex<Option<ChainedInfo>>>,
+    /// Atomic sample counter — incremented by CountingSource in the audio thread.
+    /// Progress task reads this for drift-free position tracking.
+    pub samples_played: Arc<AtomicU64>,
+    /// Sample rate of the currently playing source (for samples → seconds).
+    pub current_sample_rate: Arc<AtomicU32>,
+    /// Channel count of the currently playing source.
+    pub current_channels: Arc<AtomicU32>,
+    /// Instant (as nanos since UNIX epoch via Instant hack) of the last gapless
+    /// auto-advance. Commands arriving within 500 ms are rejected as ghost commands.
+    pub gapless_switch_at: Arc<AtomicU64>,
+    /// Active radio session state.  None for regular (non-radio) tracks.
+    /// Dropping the value aborts the HTTP download task via RadioLiveState::Drop.
+    pub(crate) radio_state: Mutex<Option<crate::audio::stream::RadioLiveState>>,
+    /// URL last committed to `AudioCurrent` — used so `audio_update_replay_gain` can
+    /// resolve LUFS / startup trim when the frontend passes `loudnessGainDb: null`
+    /// (otherwise `compute_gain` would treat that as unity gain and playback "jumps").
+    pub(crate) current_playback_url: Arc<Mutex<Option<String>>>,
+    /// Subsonic song id last passed from JS with `audio_play` (trimmed). Used
+    /// for loudness/waveform cache when the URL is `psysonic-local://…`.
+    pub(crate) current_analysis_track_id: Arc<Mutex<Option<String>>>,
+    /// While a `RangedHttpSource` download task is filling the buffer for this
+    /// `(track_id, play_generation)`, skip `analysis_enqueue_seed_from_url` for the
+    /// same id — otherwise a parallel full GET + Symphonia competes with playback
+    /// decode (ALSA underruns). The ranged task clears this on exit; `gen` avoids a
+    /// late drop clearing a newer play of the same track.
+    pub(crate) ranged_loudness_seed_hold: Arc<Mutex<Option<(String, u64)>>>,
+    /// Secondary sink dedicated to track previews. Runs on the same `OutputStream`
+    /// as the main sink (rodio mixes both internally) so we don't open a second
+    /// device handle — important on ALSA-exclusive hardware.
+    pub(crate) preview_sink: Arc<Mutex<Option<Arc<Sink>>>>,
+    /// Cancel token for the active preview. Bumped on every `audio_preview_play`
+    /// and `audio_preview_stop` so that orphan timer/progress tasks bail out.
+    pub(crate) preview_gen: Arc<AtomicU64>,
+    /// True when `audio_preview_play` paused the main sink and should resume it
+    /// on preview end. False if the main sink was already paused (or empty).
+    pub(crate) preview_main_resume: Arc<AtomicBool>,
+    /// Subsonic song id of the currently playing preview. Echoed back in
+    /// `audio:preview-end` so the frontend can clear UI state for that row.
+    pub(crate) preview_song_id: Arc<Mutex<Option<String>>>,
+}
+
+pub struct AudioCurrent {
+    pub sink: Option<Arc<Sink>>,
+    pub duration_secs: f64,
+    pub seek_offset: f64,
+    pub play_started: Option<Instant>,
+    pub paused_at: Option<f64>,
+    pub replay_gain_linear: f32,
+    pub base_volume: f32,
+    /// Crossfade: trigger for sample-level fade-out of the current source.
+    pub fadeout_trigger: Option<Arc<AtomicBool>>,
+    /// Crossfade: total fade samples (set before triggering).
+    pub fadeout_samples: Option<Arc<AtomicU64>>,
+}
+
+impl AudioCurrent {
+    pub fn position(&self) -> f64 {
+        if let Some(p) = self.paused_at {
+            return p;
+        }
+        if let Some(t) = self.play_started {
+            let elapsed = t.elapsed().as_secs_f64();
+            (self.seek_offset + elapsed).min(self.duration_secs.max(0.001))
+        } else {
+            self.seek_offset
+        }
+    }
+}
+
+/// Open an output device at `desired_rate` Hz (0 = device default).
+///
+/// `device_name`: exact name from `audio_list_devices`. `None` → system default.
+/// Falls back to the system default if the named device is not found.
+///
+/// Resolution order:
+///   1. Exact rate match in the device's supported config ranges.
+///   2. Highest available rate (for hardware that doesn't support the source rate).
+///   3. Device default.
+///   4. System default (last resort).
+///
+/// Returns `(OutputStream, OutputStreamHandle, actual_sample_rate)`.
+fn open_stream_for_device_and_rate(device_name: Option<&str>, desired_rate: u32) -> (rodio::OutputStream, rodio::OutputStreamHandle, u32) {
+    use rodio::cpal::traits::{DeviceTrait, HostTrait};
+
+    // Suppress ALSA stderr noise while enumerating devices on Unix.
+    #[cfg(unix)]
+    let _guard = unsafe {
+        struct StderrGuard(i32);
+        impl Drop for StderrGuard {
+            fn drop(&mut self) { unsafe { libc::dup2(self.0, 2); libc::close(self.0); } }
+        }
+        let saved = libc::dup(2);
+        let devnull = libc::open(b"/dev/null\0".as_ptr() as *const libc::c_char, libc::O_WRONLY);
+        libc::dup2(devnull, 2);
+        libc::close(devnull);
+        StderrGuard(saved)
+    };
+
+    let host = rodio::cpal::default_host();
+
+    // Resolve the target device: explicit name first, then (on Linux) prefer
+    // a "pipewire" or "pulse" ALSA alias before falling back to cpal's system
+    // default. On PipeWire-based distros the raw ALSA `default` alias can
+    // route to a null sink at app-start (issue #234 on Debian 13): the stream
+    // opens cleanly, progress ticks run, no audio reaches the user. The
+    // named-alias path goes through pipewire-alsa's real sink and just works.
+    // On systems where neither alias exists (pure ALSA, macOS, Windows),
+    // `find_by_name` returns None and we drop through to `default_output_device`
+    // unchanged — no regression.
+    let find_by_name = |name: &str| -> Option<_> {
+        host.output_devices().ok()?.find(|d| d.name().ok().as_deref() == Some(name))
+    };
+
+    let device = device_name
+        .and_then(find_by_name)
+        .or_else(|| {
+            #[cfg(target_os = "linux")]
+            { find_by_name("pipewire").or_else(|| find_by_name("pulse")) }
+            #[cfg(not(target_os = "linux"))]
+            { None }
+        })
+        .or_else(|| host.default_output_device());
+
+    if let Some(device) = device {
+        if desired_rate > 0 {
+            if let Ok(supported) = device.supported_output_configs() {
+                let configs: Vec<_> = supported.collect();
+
+                // 1. Exact rate match — prefer more channels (stereo > mono).
+                let exact = configs.iter()
+                    .filter(|c| {
+                        c.min_sample_rate().0 <= desired_rate
+                            && desired_rate <= c.max_sample_rate().0
+                    })
+                    .max_by_key(|c| c.channels());
+
+                if let Some(cfg) = exact {
+                    let config = cfg.clone()
+                        .with_sample_rate(rodio::cpal::SampleRate(desired_rate));
+                    if let Ok((stream, handle)) =
+                        rodio::OutputStream::try_from_device_config(&device, config)
+                    {
+                        crate::app_eprintln!("[psysonic] audio stream opened at {} Hz (exact)", desired_rate);
+                        return (stream, handle, desired_rate);
+                    }
+                }
+
+                // 2. No exact match — use the highest supported rate.
+                let best = configs.iter()
+                    .max_by_key(|c| c.max_sample_rate().0);
+
+                if let Some(cfg) = best {
+                    let rate = cfg.max_sample_rate().0;
+                    let config = cfg.clone()
+                        .with_sample_rate(rodio::cpal::SampleRate(rate));
+                    if let Ok((stream, handle)) =
+                        rodio::OutputStream::try_from_device_config(&device, config)
+                    {
+                        crate::app_eprintln!(
+                            "[psysonic] audio stream opened at {} Hz (highest, wanted {})",
+                            rate, desired_rate
+                        );
+                        return (stream, handle, rate);
+                    }
+                }
+            }
+        }
+
+        // 3. Device default.
+        if let Ok((stream, handle)) = rodio::OutputStream::try_from_device(&device) {
+            let rate = device
+                .default_output_config()
+                .map(|c| c.sample_rate().0)
+                .unwrap_or(44100);
+            crate::app_eprintln!("[psysonic] audio stream opened at {} Hz (device default)", rate);
+            return (stream, handle, rate);
+        }
+    }
+
+    // 4. Last resort: system default.
+    crate::app_eprintln!("[psysonic] audio stream falling back to system default");
+    let (stream, handle) = rodio::OutputStream::try_default()
+        .expect("cannot open any audio output device");
+    let rate = rodio::cpal::default_host()
+        .default_output_device()
+        .and_then(|d| d.default_output_config().ok())
+        .map(|c| c.sample_rate().0)
+        .unwrap_or(44100);
+    (stream, handle, rate)
+}
+
+pub fn create_engine() -> (AudioEngine, std::thread::JoinHandle<()>) {
+    // macOS: request a smaller CoreAudio buffer to reduce output latency.
+    #[cfg(target_os = "macos")]
+    {
+        if std::env::var("COREAUDIO_BUFFER_SIZE").is_err() {
+            std::env::set_var("COREAUDIO_BUFFER_SIZE", "512");
+        }
+    }
+
+    // Channels: main thread ←→ audio-stream thread.
+    //   init_tx/rx : (OutputStreamHandle, actual_rate) sent once at startup.
+    //   reopen_tx/rx: (desired_rate, reply_tx) — triggers a stream re-open.
+    let (init_tx, init_rx) =
+        std::sync::mpsc::sync_channel::<(rodio::OutputStreamHandle, u32)>(0);
+    let (reopen_tx, reopen_rx) =
+        std::sync::mpsc::sync_channel::<(u32, bool, Option<String>, std::sync::mpsc::SyncSender<rodio::OutputStreamHandle>)>(4);
+
+    let thread = std::thread::Builder::new()
+        .name("psysonic-audio-stream".into())
+        .spawn(move || {
+            // Set PipeWire / PulseAudio latency hints before the first open.
+            #[cfg(target_os = "linux")]
+            {
+                // Match cpal ALSA ~200 ms headroom: larger quantum reduces underruns when
+                // the decoder thread catches up after seek or competes with other work.
+                if std::env::var("PIPEWIRE_LATENCY").is_err() {
+                    std::env::set_var("PIPEWIRE_LATENCY", "8192/48000");
+                }
+                if std::env::var("PULSE_LATENCY_MSEC").is_err() {
+                    std::env::set_var("PULSE_LATENCY_MSEC", "170");
+                }
+            }
+
+            // Thread priority is kept at default during standard-mode playback.
+            // It is escalated to Max only when a Hi-Res stream reopen is requested,
+            // to prevent PipeWire underruns at high quantum sizes (8192 frames).
+            let (mut _stream, handle, rate) = open_stream_for_device_and_rate(None, 0);
+            init_tx.send((handle, rate)).ok();
+
+            // Keep the stream alive and handle sample-rate / device-switch requests.
+            while let Ok((desired_rate, is_hi_res, device_name, reply_tx)) = reopen_rx.recv() {
+                // Escalate to Max for Hi-Res reopens (large PipeWire quanta need
+                // real-time scheduling to avoid underruns). No escalation for
+                // standard mode — the thread blocks on recv() between reopens so
+                // elevated priority would only waste scheduler budget.
+                if is_hi_res {
+                    thread_priority::set_current_thread_priority(
+                        thread_priority::ThreadPriority::Max
+                    ).ok();
+                }
+
+                drop(_stream); // close old stream before opening new one
+
+                // Scale the PipeWire quantum with the sample rate so wall-clock
+                // latency stays roughly constant (≈93 ms) at all rates.
+                // 8192 frames at 88200 Hz ≈ 92.9 ms (same as 4096 at 48000 Hz).
+                #[cfg(target_os = "linux")]
+                {
+                    let frames: u32 = if desired_rate > 48_000 { 8192 } else { 4096 };
+                    std::env::set_var("PIPEWIRE_LATENCY", format!("{frames}/{desired_rate}"));
+                    // Keep PULSE_LATENCY_MSEC in sync so PulseAudio-based setups
+                    // get the same wall-clock quantum as PipeWire.
+                    let latency_ms = (frames as f64 / desired_rate as f64 * 1000.0).round() as u64;
+                    std::env::set_var("PULSE_LATENCY_MSEC", latency_ms.to_string());
+                }
+
+                let (new_stream, new_handle, _actual) = open_stream_for_device_and_rate(device_name.as_deref(), desired_rate);
+                _stream = new_stream;
+                reply_tx.send(new_handle).ok();
+            }
+        })
+        .expect("spawn audio stream thread");
+
+    let (initial_handle, initial_rate) = init_rx.recv().expect("audio stream handle");
+
+    let engine = AudioEngine {
+        stream_handle: Arc::new(std::sync::Mutex::new(initial_handle)),
+        stream_sample_rate: Arc::new(AtomicU32::new(initial_rate)),
+        device_default_rate: initial_rate,
+        stream_reopen_tx: reopen_tx,
+        selected_device: Arc::new(Mutex::new(None)),
+        current: Arc::new(Mutex::new(AudioCurrent {
+            sink: None,
+            duration_secs: 0.0,
+            seek_offset: 0.0,
+            play_started: None,
+            paused_at: None,
+            replay_gain_linear: 1.0,
+            base_volume: 0.8,
+            fadeout_trigger: None,
+            fadeout_samples: None,
+        })),
+        generation: Arc::new(AtomicU64::new(0)),
+        http_client: Arc::new(RwLock::new(
+            reqwest::Client::builder()
+                .timeout(Duration::from_secs(30))
+                .use_rustls_tls()
+                .user_agent(crate::subsonic_wire_user_agent())
+                .build()
+                .unwrap_or_default(),
+        )),
+        eq_gains: Arc::new(std::array::from_fn(|_| AtomicU32::new(0f32.to_bits()))),
+        eq_enabled: Arc::new(AtomicBool::new(false)),
+        eq_pre_gain: Arc::new(AtomicU32::new(0f32.to_bits())),
+        preloaded: Arc::new(Mutex::new(None)),
+        stream_completed_cache: Arc::new(Mutex::new(None)),
+        current_is_seekable: Arc::new(AtomicBool::new(true)),
+        crossfade_enabled: Arc::new(AtomicBool::new(false)),
+        crossfade_secs: Arc::new(AtomicU32::new(3.0f32.to_bits())),
+        fading_out_sink: Arc::new(Mutex::new(None)),
+        gapless_enabled: Arc::new(AtomicBool::new(false)),
+        normalization_engine: Arc::new(AtomicU32::new(0)),
+        normalization_target_lufs: Arc::new(AtomicU32::new((-16.0f32).to_bits())),
+        loudness_pre_analysis_attenuation_db: Arc::new(AtomicU32::new((-4.5f32).to_bits())),
+        chained_info: Arc::new(Mutex::new(None)),
+        samples_played: Arc::new(AtomicU64::new(0)),
+        current_sample_rate: Arc::new(AtomicU32::new(0)),
+        current_channels: Arc::new(AtomicU32::new(2)),
+        gapless_switch_at: Arc::new(AtomicU64::new(0)),
+        radio_state: Mutex::new(None),
+        current_playback_url: Arc::new(Mutex::new(None)),
+        current_analysis_track_id: Arc::new(Mutex::new(None)),
+        ranged_loudness_seed_hold: Arc::new(Mutex::new(None)),
+        preview_sink: Arc::new(Mutex::new(None)),
+        preview_gen: Arc::new(AtomicU64::new(0)),
+        preview_main_resume: Arc::new(AtomicBool::new(false)),
+        preview_song_id: Arc::new(Mutex::new(None)),
+    };
+
+    (engine, thread)
+}
+/// `analysis_enqueue_seed_from_url` should bail while this track's ranged HTTP buffer
+/// is still filling — playback will seed on completion with the same bytes.
+pub(crate) fn ranged_loudness_backfill_should_defer(engine: &AudioEngine, track_id: &str) -> bool {
+    let tid = track_id.trim();
+    if tid.is_empty() {
+        return false;
+    }
+    let Ok(g) = engine.ranged_loudness_seed_hold.lock() else {
+        return false;
+    };
+    matches!(&*g, Some((t, _)) if t.as_str() == tid)
+}
+
+/// Subsonic id pinned for the playing source (`audio_play`). Used to prioritize
+/// HTTP loudness backfill for the track the user is listening to.
+pub(crate) fn analysis_track_id_is_current_playback(engine: &AudioEngine, track_id: &str) -> bool {
+    let needle = track_id.trim();
+    if needle.is_empty() {
+        return false;
+    }
+    let Ok(guard) = engine.current_analysis_track_id.lock() else {
+        return false;
+    };
+    let Some(cur) = guard.as_deref().map(str::trim).filter(|s| !s.is_empty()) else {
+        return false;
+    };
+    cur == needle
+}
+
+pub(crate) fn audio_http_client(state: &AudioEngine) -> reqwest::Client {
+    state
+        .http_client
+        .read()
+        .map(|c| c.clone())
+        .unwrap_or_default()
+}
+
+pub fn refresh_http_user_agent(state: &AudioEngine, ua: &str) {
+    let client = reqwest::Client::builder()
+        .timeout(Duration::from_secs(30))
+        .use_rustls_tls()
+        .user_agent(ua)
+        .build()
+        .unwrap_or_default();
+    if let Ok(mut slot) = state.http_client.write() {
+        *slot = client;
+    }
+}
+pub(crate) fn analysis_seed_high_priority_for_track(app: &AppHandle, track_id: &str) -> bool {
+    app.try_state::<AudioEngine>()
+        .is_some_and(|e| analysis_track_id_is_current_playback(&e, track_id))
+}

src-tauri/src/audio/helpers.rs

@@ -0,0 +1,563 @@
+//! URL identity, loudness cache resolution, fetch, gain math, and stream analysis helpers.
+use std::sync::atomic::{AtomicU64, Ordering};
+use std::sync::Arc;
+use std::time::Duration;
+
+use futures_util::StreamExt;
+use rodio::Sink;
+use serde::Serialize;
+use tauri::{AppHandle, Emitter, Manager};
+
+use crate::audio::engine::AudioEngine;
+use crate::audio::ipc::{
+    partial_loudness_should_emit, PartialLoudnessPayload, PARTIAL_LOUDNESS_DELTA_THRESHOLD_DB,
+};
+
+pub(crate) fn emit_partial_loudness_from_bytes(
+    app: &AppHandle,
+    url: &str,
+    bytes: &[u8],
+    target_lufs: f32,
+    pre_analysis_attenuation_db: f32,
+) {
+    if bytes.len() < PARTIAL_LOUDNESS_MIN_BYTES {
+        crate::app_deprintln!(
+            "[normalization] partial-loudness skip reason=insufficient-bytes bytes={} min_bytes={}",
+            bytes.len(),
+            PARTIAL_LOUDNESS_MIN_BYTES
+        );
+        return;
+    }
+    // Lightweight fallback based on buffered bytes count to keep CPU low.
+    let mb = bytes.len() as f32 / (1024.0 * 1024.0);
+    let pre_floor = pre_analysis_attenuation_db.clamp(-24.0, 0.0);
+    // Target-derived hint (e.g. -12 LUFS → -1 dB). Old `(hint).clamp(pre, 0)` left
+    // the hint when it lay inside [pre, 0] — e.g. -1 with pre=-6, so AAC/M4A
+    // streaming often sat at -1 dB until full analysis. Combine with user trim:
+    // stricter (more negative) pre wins; milder pre still caps vs the hint.
+    let heuristic_floor = (target_lufs + 11.0).clamp(-6.0, 0.0);
+    let floor_db = if pre_floor < heuristic_floor {
+        pre_floor
+    } else {
+        pre_floor.max(heuristic_floor)
+    };
+    let gain_db = (-(mb * 0.7)).max(floor_db).min(0.0);
+    let track_key = playback_identity(url).unwrap_or_else(|| url.to_string());
+    if !partial_loudness_should_emit(&track_key, gain_db as f32) {
+        crate::app_deprintln!(
+            "[normalization] partial-loudness skip reason=delta-below-threshold gain_db={:.2} threshold_db={:.2} track_id={:?}",
+            gain_db,
+            PARTIAL_LOUDNESS_DELTA_THRESHOLD_DB,
+            playback_identity(url)
+        );
+        return;
+    }
+    crate::app_deprintln!(
+        "[normalization] partial-loudness emit bytes={} gain_db={:.2} target_lufs={:.2} track_id={:?}",
+        bytes.len(),
+        gain_db,
+        target_lufs,
+        playback_identity(url)
+    );
+    let _ = app.emit(
+        "analysis:loudness-partial",
+        PartialLoudnessPayload {
+            track_id: playback_identity(url),
+            gain_db: gain_db as f32,
+            target_lufs,
+            is_partial: true,
+        },
+    );
+}
+
+pub(crate) fn provisional_loudness_gain_from_progress(
+    downloaded: usize,
+    total_size: usize,
+    target_lufs: f32,
+    start_db_in: f32,
+) -> Option<f32> {
+    if total_size == 0 || downloaded == 0 {
+        return None;
+    }
+    let progress = (downloaded as f32 / total_size as f32).clamp(0.0, 1.0);
+    // Move from startup attenuation toward a more realistic late-stream level.
+    // This avoids staying near -2 dB and then jumping hard when final LUFS lands.
+    let start_db = start_db_in.clamp(-24.0, 0.0).min(0.0);
+    let end_db = (target_lufs + 6.0).clamp(-10.0, -3.0).min(0.0);
+    let shaped = progress.powf(0.75);
+    Some(start_db + (end_db - start_db) * shaped)
+}
+
+pub(crate) fn content_type_to_hint(ct: &str) -> Option<String> {
+    let ct = ct.to_ascii_lowercase();
+    if ct.contains("mpeg") || ct.contains("mp3") { Some("mp3".into()) }
+    else if ct.contains("aac") || ct.contains("aacp") { Some("aac".into()) }
+    else if ct.contains("ogg") { Some("ogg".into()) }
+    else if ct.contains("flac") { Some("flac".into()) }
+    else if ct.contains("wav") || ct.contains("wave") { Some("wav".into()) }
+    else if ct.contains("opus") { Some("opus".into()) }
+    else { None }
+}
+// ─── Event payloads ───────────────────────────────────────────────────────────
+
+#[derive(Clone, Serialize)]
+pub struct ProgressPayload {
+    pub current_time: f64,
+    pub duration: f64,
+}
+
+// ─── Helpers ──────────────────────────────────────────────────────────────────
+
+/// Subsonic `buildStreamUrl()` uses a fresh random salt on every call, so two
+/// URLs for the same track differ in `t`/`s` query params. Compare a stable key.
+pub(crate) fn playback_identity(url: &str) -> Option<String> {
+    if let Some(path) = url.strip_prefix("psysonic-local://") {
+        return Some(format!("local:{path}"));
+    }
+    if !url.contains("stream.view") {
+        return None;
+    }
+    let q = url.split('?').nth(1)?;
+    for pair in q.split('&') {
+        if let Some(v) = pair.strip_prefix("id=") {
+            let v = v.split('&').next().unwrap_or(v);
+            return Some(format!("stream:{v}"));
+        }
+    }
+    None
+}
+
+/// Stable id for analysis cache rows and `analysis:waveform-updated`.
+/// Prefer the Subsonic track id from the frontend: `psysonic-local://` URLs
+/// only map to `local:path` in `playback_identity`, which does not match
+/// `analysis_get_waveform_for_track(trackId)` or the UI's `currentTrack.id`.
+pub(crate) fn analysis_cache_track_id(logical_track_id: Option<&str>, url: &str) -> Option<String> {
+    let logical = logical_track_id
+        .map(str::trim)
+        .filter(|s| !s.is_empty())
+        .map(|s| s.to_string());
+    logical.or_else(|| playback_identity(url))
+}
+
+pub(crate) fn same_playback_target(a_url: &str, b_url: &str) -> bool {
+    match (playback_identity(a_url), playback_identity(b_url)) {
+        (Some(a), Some(b)) => a == b,
+        _ => a_url == b_url,
+    }
+}
+
+#[derive(Clone, Copy)]
+pub(crate) struct ResolveLoudnessCacheOpts {
+    /// When false, skip `get_latest_waveform_for_track` — `audio_update_replay_gain` runs
+    /// on every partial-LUFS tick; loudness gain does not depend on waveform, and the extra
+    /// SQLite read was pure overhead on the IPC path.
+    pub(crate) touch_waveform: bool,
+    /// When false, omit `cache-miss` / `cache-invalid` debug lines (still log hits and errors).
+    pub(crate) log_soft_misses: bool,
+}
+
+impl Default for ResolveLoudnessCacheOpts {
+    fn default() -> Self {
+        Self {
+            touch_waveform: true,
+            log_soft_misses: true,
+        }
+    }
+}
+
+pub(crate) fn resolve_loudness_gain_from_cache(
+    app: &AppHandle,
+    url: &str,
+    target_lufs: f32,
+    logical_track_id: Option<&str>,
+) -> Option<f32> {
+    resolve_loudness_gain_from_cache_impl(
+        app,
+        url,
+        target_lufs,
+        logical_track_id,
+        ResolveLoudnessCacheOpts::default(),
+    )
+}
+
+pub(crate) fn resolve_loudness_gain_from_cache_impl(
+    app: &AppHandle,
+    url: &str,
+    target_lufs: f32,
+    logical_track_id: Option<&str>,
+    opts: ResolveLoudnessCacheOpts,
+) -> Option<f32> {
+    // Only a SQLite loudness row counts here. Ephemeral JS hints (`analysis:loudness-partial`)
+    // are applied in `audio_update_replay_gain` via `loudness_gain_db_or_startup(..., true, _)`.
+    let Some(track_id) = analysis_cache_track_id(logical_track_id, url) else {
+        if opts.log_soft_misses {
+            crate::app_deprintln!(
+                "[normalization] resolve_loudness_gain source=no-identity url_len={}",
+                url.len()
+            );
+        }
+        return None;
+    };
+    let Some(cache) = app.try_state::<crate::analysis_cache::AnalysisCache>() else {
+        if opts.log_soft_misses {
+            crate::app_deprintln!(
+                "[normalization] resolve_loudness_gain source=no-analysis-cache track_id={}",
+                track_id
+            );
+        }
+        return None;
+    };
+    if opts.touch_waveform {
+        // Bind / preload: verify waveform context exists alongside loudness lookup.
+        let _ = cache.get_latest_waveform_for_track(&track_id);
+    }
+    match cache.get_latest_loudness_for_track(&track_id) {
+        Ok(Some(row)) if row.integrated_lufs.is_finite() => {
+            let recommended = crate::analysis_cache::recommended_gain_for_target(
+                row.integrated_lufs,
+                row.true_peak,
+                target_lufs as f64,
+            ) as f32;
+            crate::app_deprintln!(
+                "[normalization] resolve_loudness_gain source=cache track_id={} gain_db={:.2} target_lufs={:.2} integrated_lufs={:.2} updated_at={}",
+                track_id,
+                recommended,
+                target_lufs,
+                row.integrated_lufs,
+                row.updated_at
+            );
+            Some(recommended)
+        }
+        Ok(Some(row)) => {
+            if opts.log_soft_misses {
+                crate::app_deprintln!(
+                    "[normalization] resolve_loudness_gain source=cache-invalid track_id={} integrated_lufs={}",
+                    track_id,
+                    row.integrated_lufs
+                );
+            }
+            None
+        }
+        Ok(None) => {
+            if opts.log_soft_misses {
+                crate::app_deprintln!(
+                    "[normalization] resolve_loudness_gain source=cache-miss track_id={}",
+                    track_id
+                );
+            }
+            None
+        }
+        Err(e) => {
+            crate::app_deprintln!(
+                "[normalization] resolve_loudness_gain source=cache-error track_id={} err={}",
+                track_id,
+                e
+            );
+            None
+        }
+    }
+}
+
+/// Typical integrated LUFS (streaming pivot) when SQLite has no row yet — so target changes
+/// still move gain before real analysis completes.
+const LOUDNESS_PLACEHOLDER_INTEGRATED_LUFS: f64 = -14.0;
+
+#[inline]
+pub(crate) fn loudness_gain_placeholder_until_cache(target_lufs: f32, pre_analysis_attenuation_db: f32) -> f32 {
+    let pre = pre_analysis_attenuation_db.clamp(-24.0, 0.0).min(0.0);
+    // `true_peak = 0.0` skips the headroom cap until integrated measurement exists.
+    let pivot = crate::analysis_cache::recommended_gain_for_target(
+        LOUDNESS_PLACEHOLDER_INTEGRATED_LUFS,
+        0.0,
+        f64::from(target_lufs),
+    ) as f32;
+    (pivot + pre).clamp(-24.0, 24.0)
+}
+
+/// LUFS gain after a single `resolve_loudness_gain_from_cache` result (`None` = miss).
+/// Keeps `audio_update_replay_gain` / `audio_play` from resolving twice on the same URL.
+/// Until a cache row exists, follow current target (see [`loudness_gain_placeholder_until_cache`]).
+pub(crate) fn loudness_gain_db_after_resolve(
+    resolved_from_cache: Option<f32>,
+    target_lufs: f32,
+    pre_analysis_attenuation_db: f32,
+    allow_js_when_uncached: bool,
+    js_gain_db: Option<f32>,
+) -> Option<f32> {
+    let uncached = loudness_gain_placeholder_until_cache(target_lufs, pre_analysis_attenuation_db);
+    match resolved_from_cache {
+        Some(g) => Some(g),
+        None => {
+            if allow_js_when_uncached {
+                match js_gain_db {
+                    Some(r) if r.is_finite() => Some(r),
+                    _ => Some(uncached),
+                }
+            } else {
+                Some(uncached)
+            }
+        }
+    }
+}
+
+/// LUFS: DB-backed integrated LUFS only at bind time (`allow_js_when_uncached = false`);
+/// after `analysis:loudness-partial`, `audio_update_replay_gain` passes `true` so finite
+/// JS gain applies until SQLite catches up. Must never return `None` or `compute_gain` uses unity.
+pub(crate) fn loudness_gain_db_or_startup(
+    app: &AppHandle,
+    url: &str,
+    target_lufs: f32,
+    logical_track_id: Option<&str>,
+    pre_analysis_attenuation_db: f32,
+    allow_js_when_uncached: bool,
+    js_gain_db: Option<f32>,
+) -> Option<f32> {
+    let resolved = resolve_loudness_gain_from_cache_impl(
+        app,
+        url,
+        target_lufs,
+        logical_track_id,
+        ResolveLoudnessCacheOpts::default(),
+    );
+    loudness_gain_db_after_resolve(
+        resolved,
+        target_lufs,
+        pre_analysis_attenuation_db,
+        allow_js_when_uncached,
+        js_gain_db,
+    )
+}
+
+#[inline]
+pub(crate) fn loudness_pre_analysis_db_for_engine(state: &AudioEngine) -> f32 {
+    f32::from_bits(
+        state
+            .loudness_pre_analysis_attenuation_db
+            .load(Ordering::Relaxed),
+    )
+    .clamp(-24.0, 0.0)
+    .min(0.0)
+}
+
+/// Take (consume) completed manual-stream bytes if they correspond to `url`.
+pub fn take_stream_completed_for_url(state: &AudioEngine, url: &str) -> Option<Vec<u8>> {
+    let mut guard = state.stream_completed_cache.lock().unwrap();
+    if guard
+        .as_ref()
+        .is_some_and(|p| same_playback_target(&p.url, url))
+    {
+        return guard.take().map(|p| p.data);
+    }
+    None
+}
+
+/// Fetch track bytes from the preload cache or via HTTP.
+pub(crate) async fn fetch_data(
+    url: &str,
+    state: &AudioEngine,
+    gen: u64,
+    app: &AppHandle,
+) -> Result<Option<Vec<u8>>, String> {
+    // Check completed streamed-track cache first (manual streaming fallback cache).
+    let streamed_cached = {
+        let mut streamed = state.stream_completed_cache.lock().unwrap();
+        if streamed.as_ref().is_some_and(|p| same_playback_target(&p.url, url)) {
+            streamed.take().map(|p| p.data)
+        } else {
+            None
+        }
+    };
+    if let Some(data) = streamed_cached {
+        return Ok(Some(data));
+    }
+
+    // Check preload cache next.
+    let cached = {
+        let mut preloaded = state.preloaded.lock().unwrap();
+        if preloaded.as_ref().is_some_and(|p| same_playback_target(&p.url, url)) {
+            preloaded.take().map(|p| p.data)
+        } else {
+            None
+        }
+    };
+
+    if let Some(data) = cached {
+        return Ok(Some(data));
+    }
+
+    // Offline cache — local file written by download_track_offline.
+    if let Some(path) = url.strip_prefix("psysonic-local://") {
+        let data = tokio::fs::read(path).await.map_err(|e| e.to_string())?;
+        return Ok(Some(data));
+    }
+
+    let response = crate::audio::engine::audio_http_client(&state).get(url).send().await.map_err(|e| e.to_string())?;
+    let status = response.status();
+    let ct = response.headers()
+        .get(reqwest::header::CONTENT_TYPE)
+        .and_then(|v| v.to_str().ok())
+        .unwrap_or("-");
+    let server_hdr = response.headers()
+        .get("server")
+        .and_then(|v| v.to_str().ok())
+        .unwrap_or("-");
+    // Strip auth params from URL before logging.
+    let safe_url = url.split('?').next().unwrap_or(url);
+    crate::app_deprintln!(
+        "[audio] fetch {} → {} | content-type: {} | server: {}",
+        safe_url, status, ct, server_hdr
+    );
+    if !response.status().is_success() {
+        if state.generation.load(Ordering::SeqCst) != gen {
+            return Ok(None); // superseded
+        }
+        let status = response.status().as_u16();
+        let msg = format!("HTTP {status}");
+        app.emit("audio:error", &msg).ok();
+        return Err(msg);
+    }
+    // Stream the body, checking gen between chunks so a rapid manual skip can
+    // abort a superseded download mid-flight and free bandwidth for the new one.
+    let hint = response.content_length().unwrap_or(0) as usize;
+    let mut stream = response.bytes_stream();
+    let mut data = Vec::with_capacity(hint);
+    while let Some(chunk) = stream.next().await {
+        if state.generation.load(Ordering::SeqCst) != gen {
+            return Ok(None); // superseded — abort
+        }
+        data.extend_from_slice(&chunk.map_err(|e| e.to_string())?);
+    }
+    Ok(Some(data))
+}
+
+/// When playback uses full track bytes already in RAM (gapless `reuse_chained_bytes`,
+/// `preloaded`, or `stream_completed_cache` via `fetch_data`), the `psysonic-local`
+/// disk-read seed path never runs. Submit the same full-buffer analysis via the cpu-seed queue so waveform /
+/// loudness SQLite can fill **offline** without `analysis_enqueue_seed_from_url` HTTP.
+pub(crate) fn spawn_analysis_seed_from_in_memory_bytes(
+    app: &AppHandle,
+    cache_track_id: Option<&str>,
+    gen: u64,
+    gen_arc: &Arc<AtomicU64>,
+    bytes: &[u8],
+) {
+    let Some(track_id) = cache_track_id.map(str::trim).filter(|s| !s.is_empty()) else {
+        return;
+    };
+    if bytes.is_empty() || bytes.len() > crate::audio::stream::TRACK_STREAM_PROMOTE_MAX_BYTES {
+        return;
+    }
+    let track_id = track_id.to_string();
+    let bytes = bytes.to_vec();
+    let app = app.clone();
+    let gen_arc = gen_arc.clone();
+    crate::app_deprintln!(
+        "[stream] in-memory play path: scheduling full-track analysis track_id={} size_mib={:.2}",
+        track_id,
+        bytes.len() as f64 / (1024.0 * 1024.0)
+    );
+    let high = crate::audio::engine::analysis_seed_high_priority_for_track(&app, &track_id);
+    tokio::spawn(async move {
+        if gen_arc.load(Ordering::SeqCst) != gen {
+            return;
+        }
+        if let Err(e) = crate::submit_analysis_cpu_seed(app.clone(), track_id.clone(), bytes, high).await {
+            crate::app_eprintln!(
+                "[analysis] in-memory play path seed failed for {}: {}",
+                track_id,
+                e
+            );
+        }
+    });
+}
+
+/// -1 dB headroom applied at full scale to prevent inter-sample clipping.
+/// Modern masters are often at 0 dBFS; the EQ biquad chain and resampler
+/// can produce inter-sample peaks slightly above ±1.0 → audible distortion.
+/// 10^(-1/20) ≈ 0.891 — inaudible volume difference, eliminates clipping.
+pub(crate) const MASTER_HEADROOM: f32 = 0.891_254;
+pub(crate) const PARTIAL_LOUDNESS_MIN_BYTES: usize = 256 * 1024;
+pub(crate) const PARTIAL_LOUDNESS_EMIT_INTERVAL_MS: u64 = 900;
+
+pub(crate) fn compute_gain(
+    normalization_engine: u32,
+    replay_gain_db: Option<f32>,
+    replay_gain_peak: Option<f32>,
+    loudness_gain_db: Option<f32>,
+    pre_gain_db: f32,
+    fallback_db: f32,
+    volume: f32,
+) -> (f32, f32) {
+    let gain_linear = match normalization_engine {
+        2 => loudness_gain_db
+            .map(|db| 10f32.powf(db / 20.0))
+            .unwrap_or(1.0),
+        1 => replay_gain_db
+            .map(|db| 10f32.powf((db + pre_gain_db) / 20.0))
+            .unwrap_or_else(|| 10f32.powf(fallback_db / 20.0)),
+        _ => 1.0,
+    };
+    let peak = if normalization_engine == 1 {
+        replay_gain_peak.unwrap_or(1.0).max(0.001)
+    } else {
+        1.0
+    };
+    let gain_linear = gain_linear.min(1.0 / peak);
+    let effective = (volume.clamp(0.0, 1.0) * gain_linear * MASTER_HEADROOM).clamp(0.0, 1.0);
+    (gain_linear, effective)
+}
+
+pub(crate) fn normalization_engine_name(mode: u32) -> &'static str {
+    match mode {
+        1 => "replaygain",
+        2 => "loudness",
+        _ => "off",
+    }
+}
+
+pub(crate) fn gain_linear_to_db(gain_linear: f32) -> Option<f32> {
+    if gain_linear.is_finite() && gain_linear > 0.0 {
+        Some(20.0 * gain_linear.log10())
+    } else {
+        None
+    }
+}
+
+/// `audio:normalization-state` “Now dB” for the UI: effective applied gain, including
+/// loudness pre-analysis trim from settings when no cache row exists yet (matches audible level).
+pub(crate) fn loudness_ui_current_gain_db(gain_linear: f32) -> Option<f32> {
+    gain_linear_to_db(gain_linear)
+}
+
+pub(crate) fn ramp_sink_volume(sink: Arc<Sink>, from: f32, to: f32) {
+    let from = from.clamp(0.0, 1.0);
+    let to = to.clamp(0.0, 1.0);
+    if (to - from).abs() < 0.002 {
+        sink.set_volume(to);
+        return;
+    }
+    static RAMP_GEN: AtomicU64 = AtomicU64::new(0);
+    let my_gen = RAMP_GEN.fetch_add(1, Ordering::SeqCst) + 1;
+    std::thread::spawn(move || {
+        let delta = (to - from).abs();
+        // Stretch large corrections to avoid audible "step down" moments.
+        let (steps, step_ms): (usize, u64) = if delta > 0.30 {
+            (24, 35)
+        } else if delta > 0.18 {
+            (18, 30)
+        } else if delta > 0.10 {
+            (14, 24)
+        } else {
+            (8, 16)
+        };
+        for i in 1..=steps {
+            if RAMP_GEN.load(Ordering::SeqCst) != my_gen {
+                return;
+            }
+            let t = i as f32 / steps as f32;
+            let v = from + (to - from) * t;
+            sink.set_volume(v.clamp(0.0, 1.0));
+            std::thread::sleep(Duration::from_millis(step_ms));
+        }
+    });
+}

src-tauri/src/audio/ipc.rs

@@ -0,0 +1,76 @@
+//! Deduped emits for normalization UI and partial loudness analysis.
+use serde::Serialize;
+use std::sync::{Mutex, OnceLock};
+use tauri::{AppHandle, Emitter};
+
+#[derive(Clone, Serialize)]
+#[serde(rename_all = "camelCase")]
+pub(crate) struct PartialLoudnessPayload {
+    pub(crate) track_id: Option<String>,
+    pub(crate) gain_db: f32,
+    pub(crate) target_lufs: f32,
+    pub(crate) is_partial: bool,
+}
+
+#[derive(Clone, Serialize)]
+#[serde(rename_all = "camelCase")]
+pub(crate) struct NormalizationStatePayload {
+    pub(crate) engine: String,
+    pub(crate) current_gain_db: Option<f32>,
+    pub(crate) target_lufs: f32,
+}
+
+/// Last `audio:normalization-state` emit, kept so we can suppress duplicate
+/// payloads. The frontend already debounces this event, but on Windows
+/// (WebView2) the IPC pipe is the bottleneck — every echo we skip here is
+/// renderer-thread time we don't pay.
+pub(crate) static LAST_NORM_STATE_EMIT: OnceLock<Mutex<Option<NormalizationStatePayload>>> = OnceLock::new();
+
+pub(crate) fn norm_state_lock() -> &'static Mutex<Option<NormalizationStatePayload>> {
+    LAST_NORM_STATE_EMIT.get_or_init(|| Mutex::new(None))
+}
+
+pub(crate) fn norm_state_changed(prev: &NormalizationStatePayload, next: &NormalizationStatePayload) -> bool {
+    if prev.engine != next.engine { return true; }
+    if (prev.target_lufs - next.target_lufs).abs() >= 0.02 { return true; }
+    match (prev.current_gain_db, next.current_gain_db) {
+        (None, None) => false,
+        (Some(a), Some(b)) => (a - b).abs() >= 0.05,
+        _ => true, // None ↔ Some transition is significant
+    }
+}
+
+pub(crate) fn maybe_emit_normalization_state(app: &AppHandle, payload: NormalizationStatePayload) {
+    let mut guard = norm_state_lock().lock().unwrap();
+    let should_emit = match guard.as_ref() {
+        Some(prev) => norm_state_changed(prev, &payload),
+        None => true,
+    };
+    if !should_emit { return; }
+    *guard = Some(payload.clone());
+    drop(guard);
+    let _ = app.emit("audio:normalization-state", payload);
+}
+
+/// Last `analysis:loudness-partial` gain emitted per track-identity, used to
+/// suppress emits whose gain hasn't moved meaningfully (≥ 0.1 dB). The partial
+/// heuristic in `emit_partial_loudness_from_bytes` and the ranged-progress curve
+/// both produce values that drift by hundredths of a dB even on identical input,
+/// so the time-based throttle alone is not enough to keep the loop quiet.
+pub(crate) static LAST_PARTIAL_LOUDNESS_EMIT: OnceLock<Mutex<std::collections::HashMap<String, f32>>> = OnceLock::new();
+pub(crate) const PARTIAL_LOUDNESS_DELTA_THRESHOLD_DB: f32 = 0.1;
+
+pub(crate) fn partial_loudness_should_emit(track_key: &str, gain_db: f32) -> bool {
+    let mut guard = LAST_PARTIAL_LOUDNESS_EMIT
+        .get_or_init(|| Mutex::new(std::collections::HashMap::new()))
+        .lock()
+        .unwrap();
+    let prev = guard.get(track_key).copied();
+    if let Some(p) = prev {
+        if (p - gain_db).abs() < PARTIAL_LOUDNESS_DELTA_THRESHOLD_DB {
+            return false;
+        }
+    }
+    guard.insert(track_key.to_string(), gain_db);
+    true
+}

src-tauri/src/audio/mod.rs

@@ -0,0 +1,24 @@
+//! Audio playback: Symphonia decode, rodio output, HTTP radio/streaming, gapless, previews.
+//!
+//! Implementation is split into submodules (`sources`, `decode`, `stream`, `commands`, …)
+//! for navigation; behavior matches the historical single `audio.rs` file.
+
+mod codec;
+pub mod commands;
+mod decode;
+mod dev_io;
+mod device_watcher;
+mod engine;
+mod helpers;
+mod ipc;
+pub mod preview;
+mod sources;
+mod state;
+mod stream;
+
+pub use commands::{audio_default_output_device_name, audio_list_devices_for_engine};
+pub use device_watcher::start_device_watcher;
+pub use engine::{create_engine, refresh_http_user_agent, AudioEngine};
+pub use helpers::take_stream_completed_for_url;
+
+pub(crate) use engine::{analysis_track_id_is_current_playback, ranged_loudness_backfill_should_defer};

src-tauri/src/audio/preview.rs

@@ -0,0 +1,301 @@
+//! Short preview playback on a secondary sink (same output stream).
+use std::sync::atomic::Ordering;
+use std::sync::Arc;
+use std::time::{Duration, Instant};
+
+use rodio::{Sink, Source};
+use tauri::{AppHandle, Emitter, State};
+
+use super::decode::SizedDecoder;
+use super::engine::{audio_http_client, AudioEngine};
+use super::helpers::MASTER_HEADROOM;
+
+// ────────────────────────────────────────────────────────────────────────────
+// Preview engine — secondary Sink on the same OutputStream, fed by Symphonia.
+// ────────────────────────────────────────────────────────────────────────────
+
+#[derive(Clone, serde::Serialize)]
+struct PreviewProgressPayload {
+    id: String,
+    elapsed: f64,
+    duration: f64,
+}
+
+#[derive(Clone, serde::Serialize)]
+struct PreviewEndPayload {
+    id: String,
+    /// "natural" = 30 s timer / source ended; "user" = explicit stop;
+    /// "interrupted" = a new preview superseded this one.
+    reason: &'static str,
+}
+
+/// Pause main sink and remember whether to resume it after preview ends.
+/// Mirrors `audio_pause` semantics so progress timestamps stay consistent.
+pub(crate) fn preview_pause_main(state: &AudioEngine) {
+    let mut cur = state.current.lock().unwrap();
+    if let Some(sink) = &cur.sink {
+        if !sink.is_paused() && !sink.empty() {
+            let pos = cur.position();
+            sink.pause();
+            cur.paused_at = Some(pos);
+            cur.play_started = None;
+            state.preview_main_resume.store(true, Ordering::Release);
+        } else {
+            state.preview_main_resume.store(false, Ordering::Release);
+        }
+    } else {
+        state.preview_main_resume.store(false, Ordering::Release);
+    }
+}
+
+/// Cancel any active preview and clear the resume marker. Called from every
+/// command that brings the main sink back to life under its own steam
+/// (`audio_play`, `audio_play_radio`, `audio_resume`) — without this the
+/// preview would keep playing in parallel and the watchdog would later try
+/// to resume a main sink that's already running, double-mixing the audio.
+pub(crate) fn preview_clear_for_new_main_playback(state: &AudioEngine, app: &AppHandle) {
+    // Order matters: clear the resume marker BEFORE bumping the generation
+    // so the watchdog — if it wakes between our writes — sees no work to do
+    // and bails without resuming main behind our back.
+    state.preview_main_resume.store(false, Ordering::Release);
+    state.preview_gen.fetch_add(1, Ordering::SeqCst);
+    let sink = state.preview_sink.lock().unwrap().take();
+    let id = state.preview_song_id.lock().unwrap().take();
+    if let Some(s) = sink { s.stop(); }
+    if let Some(id) = id {
+        app.emit("audio:preview-end", PreviewEndPayload {
+            id,
+            reason: "interrupted",
+        }).ok();
+    }
+}
+
+/// Resume main sink iff `preview_pause_main` paused it. No-op if main was
+/// already paused/empty before preview started.
+pub(crate) fn preview_resume_main(state: &AudioEngine) {
+    if !state.preview_main_resume.swap(false, Ordering::AcqRel) {
+        return;
+    }
+    let mut cur = state.current.lock().unwrap();
+    if let Some(sink) = &cur.sink {
+        if sink.is_paused() {
+            let pos = cur.paused_at.unwrap_or(cur.seek_offset);
+            sink.play();
+            cur.seek_offset = pos;
+            cur.play_started = Some(Instant::now());
+            cur.paused_at = None;
+        }
+    }
+}
+
+/// Format hint inferred from a Subsonic stream URL. The frontend always passes
+/// a `format=flac` query param for `.opus` files (server transcodes); for
+/// everything else we guess from the URL's `format=` value or fall back to None.
+pub(crate) fn preview_format_hint_from_url(url: &str) -> Option<String> {
+    url.split('?')
+        .nth(1)?
+        .split('&')
+        .find_map(|kv| {
+            let (k, v) = kv.split_once('=')?;
+            if k.eq_ignore_ascii_case("format") { Some(v.to_string()) } else { None }
+        })
+}
+
+#[tauri::command]
+pub async fn audio_preview_play(
+    id: String,
+    url: String,
+    start_sec: f64,
+    duration_sec: f64,
+    volume: f32,
+    app: AppHandle,
+    state: State<'_, AudioEngine>,
+) -> Result<(), String> {
+    let gen = state.preview_gen.fetch_add(1, Ordering::SeqCst) + 1;
+
+    // Tear down any existing preview before pausing main (so a rapid preview
+    // swap doesn't double-pause and double-resume the main sink).
+    let prev_sink = state.preview_sink.lock().unwrap().take();
+    let prev_id = state.preview_song_id.lock().unwrap().take();
+    if let Some(s) = prev_sink { s.stop(); }
+    if let Some(prev) = prev_id {
+        app.emit("audio:preview-end", PreviewEndPayload {
+            id: prev,
+            reason: "interrupted",
+        }).ok();
+    }
+
+    // Pause main if and only if we don't already hold a "main was playing"
+    // marker from a superseded preview. swap_or-style: only pause if the flag
+    // is currently false.
+    if !state.preview_main_resume.load(Ordering::Acquire) {
+        preview_pause_main(&state);
+    }
+
+    // ── Download ─────────────────────────────────────────────────────────────
+    let bytes = audio_http_client(&state)
+        .get(&url)
+        .send()
+        .await
+        .map_err(|e| format!("preview: connection failed: {e}"))?
+        .error_for_status()
+        .map_err(|e| format!("preview: HTTP {e}"))?
+        .bytes()
+        .await
+        .map_err(|e| format!("preview: read body: {e}"))?
+        .to_vec();
+
+    if state.preview_gen.load(Ordering::SeqCst) != gen {
+        // A newer preview started while we were downloading — bail.
+        return Ok(());
+    }
+
+    // ── Decode ───────────────────────────────────────────────────────────────
+    let hint = preview_format_hint_from_url(&url);
+    let bytes_for_blocking = bytes;
+    let hint_for_blocking = hint.clone();
+    let decoder = tokio::task::spawn_blocking(move || {
+        SizedDecoder::new(bytes_for_blocking, hint_for_blocking.as_deref(), false)
+    })
+    .await
+    .map_err(|e| format!("preview: decoder thread: {e}"))??;
+
+    if state.preview_gen.load(Ordering::SeqCst) != gen { return Ok(()); }
+
+    // ── Build source pipeline ────────────────────────────────────────────────
+    // Minimal pipeline: f32 conversion + take_duration cap. No EQ, no crossfade,
+    // no ReplayGain — preview is intentionally simple. Volume is set on the Sink.
+    let dur = Duration::from_secs_f64(duration_sec.max(1.0).min(120.0));
+    let source = decoder
+        .convert_samples::<f32>()
+        .take_duration(dur);
+
+    // ── Build secondary sink on the existing OutputStream ────────────────────
+    let sink = Arc::new(
+        Sink::try_new(&*state.stream_handle.lock().unwrap())
+            .map_err(|e| format!("preview: sink new: {e}"))?,
+    );
+    sink.set_volume((volume.clamp(0.0, 1.0) * MASTER_HEADROOM).clamp(0.0, 1.0));
+    sink.append(source);
+
+    // ── Best-effort seek to mid-track start position ─────────────────────────
+    // Symphonia FLAC without SEEKTABLE may fail here; preview then plays from 0
+    // which is acceptable. Hard timeout 800 ms via a worker thread.
+    if start_sec > 0.5 {
+        let start_dur = Duration::from_secs_f64(start_sec);
+        let seek_sink = sink.clone();
+        let (tx, rx) = std::sync::mpsc::channel::<()>();
+        std::thread::spawn(move || {
+            seek_sink.try_seek(start_dur).ok();
+            let _ = tx.send(());
+        });
+        let _ = rx.recv_timeout(Duration::from_millis(800));
+    }
+
+    *state.preview_sink.lock().unwrap() = Some(sink.clone());
+    *state.preview_song_id.lock().unwrap() = Some(id.clone());
+
+    app.emit("audio:preview-start", id.clone()).ok();
+
+    // ── Spawn watchdog: progress emits + auto-end ────────────────────────────
+    let preview_gen_arc = state.preview_gen.clone();
+    let preview_sink_arc = state.preview_sink.clone();
+    let preview_song_arc = state.preview_song_id.clone();
+    let preview_resume_arc = state.preview_main_resume.clone();
+    let main_current = state.current.clone();
+    let app_for_task = app.clone();
+    let id_for_task = id.clone();
+    tokio::spawn(async move {
+        let started = Instant::now();
+        let mut last_emit = Instant::now() - Duration::from_millis(300);
+        loop {
+            tokio::time::sleep(Duration::from_millis(100)).await;
+            // Cancel: another preview started or audio_preview_stop bumped the gen.
+            if preview_gen_arc.load(Ordering::SeqCst) != gen { return; }
+
+            let elapsed = started.elapsed().as_secs_f64();
+            let dur_secs = dur.as_secs_f64();
+
+            if last_emit.elapsed() >= Duration::from_millis(250) {
+                last_emit = Instant::now();
+                app_for_task.emit("audio:preview-progress", PreviewProgressPayload {
+                    id: id_for_task.clone(),
+                    elapsed: elapsed.min(dur_secs),
+                    duration: dur_secs,
+                }).ok();
+            }
+
+            // Natural end: timer expired OR sink drained early (decode error,
+            // short track, etc.).
+            let drained = match preview_sink_arc.lock().unwrap().as_ref() {
+                Some(s) => s.empty(),
+                None => true,
+            };
+            if elapsed >= dur_secs || drained {
+                // Re-check generation under the cleanup lock to avoid racing
+                // a fresh preview that bumped the counter.
+                if preview_gen_arc.load(Ordering::SeqCst) != gen { return; }
+                if let Some(s) = preview_sink_arc.lock().unwrap().take() { s.stop(); }
+                let cleared_id = preview_song_arc.lock().unwrap().take()
+                    .unwrap_or_else(|| id_for_task.clone());
+
+                // Resume main if we paused it.
+                if preview_resume_arc.swap(false, Ordering::AcqRel) {
+                    let mut cur = main_current.lock().unwrap();
+                    if let Some(sink) = &cur.sink {
+                        if sink.is_paused() {
+                            let pos = cur.paused_at.unwrap_or(cur.seek_offset);
+                            sink.play();
+                            cur.seek_offset = pos;
+                            cur.play_started = Some(Instant::now());
+                            cur.paused_at = None;
+                        }
+                    }
+                }
+
+                app_for_task.emit("audio:preview-end", PreviewEndPayload {
+                    id: cleared_id,
+                    reason: "natural",
+                }).ok();
+                return;
+            }
+        }
+    });
+
+    Ok(())
+}
+
+#[tauri::command]
+pub fn audio_preview_stop(app: AppHandle, state: State<'_, AudioEngine>) {
+    preview_stop_inner(&app, &state, true);
+}
+
+/// Like `audio_preview_stop` but leaves the main sink paused even if it had
+/// been paused by `preview_pause_main`. Used by the player-bar Stop button so
+/// "stop everything" actually goes silent — without this the engine would
+/// auto-resume main playback the moment the preview ends and the user perceives
+/// the click as having no effect.
+#[tauri::command]
+pub fn audio_preview_stop_silent(app: AppHandle, state: State<'_, AudioEngine>) {
+    preview_stop_inner(&app, &state, false);
+}
+
+pub(crate) fn preview_stop_inner(app: &AppHandle, state: &AudioEngine, resume_main: bool) {
+    state.preview_gen.fetch_add(1, Ordering::SeqCst);
+    let sink = state.preview_sink.lock().unwrap().take();
+    let id = state.preview_song_id.lock().unwrap().take();
+    if let Some(s) = sink { s.stop(); }
+
+    if resume_main {
+        preview_resume_main(state);
+    } else {
+        state.preview_main_resume.store(false, Ordering::Release);
+    }
+
+    if let Some(id) = id {
+        app.emit("audio:preview-end", PreviewEndPayload {
+            id,
+            reason: "user",
+        }).ok();
+    }
+}

src-tauri/src/audio/sources.rs

@@ -0,0 +1,401 @@
+//! Rodio `Source` wrappers: EQ, type erasure, fades, end-of-source notify, sample counter.
+use std::sync::atomic::{AtomicBool, AtomicU32, AtomicU64, Ordering};
+use std::sync::Arc;
+use std::time::Duration;
+
+use biquad::{Biquad, Coefficients, DirectForm2Transposed, ToHertz, Type as FilterType};
+use rodio::Source;
+
+// ─── 10-Band Graphic Equalizer ────────────────────────────────────────────────
+
+const EQ_BANDS_HZ: [f32; 10] = [31.0, 62.0, 125.0, 250.0, 500.0, 1000.0, 2000.0, 4000.0, 8000.0, 16000.0];
+const EQ_Q: f32 = 1.41;
+const EQ_CHECK_INTERVAL: usize = 1024;
+
+pub(crate) struct EqSource<S: Source<Item = f32>> {
+    inner: S,
+    sample_rate: u32,
+    channels: u16,
+    gains: Arc<[AtomicU32; 10]>,
+    enabled: Arc<AtomicBool>,
+    pre_gain: Arc<AtomicU32>,
+    filters: [[DirectForm2Transposed<f32>; 2]; 10],
+    current_gains: [f32; 10],
+    sample_counter: usize,
+    channel_idx: usize,
+}
+
+impl<S: Source<Item = f32>> EqSource<S> {
+    pub(crate) fn new(inner: S, gains: Arc<[AtomicU32; 10]>, enabled: Arc<AtomicBool>, pre_gain: Arc<AtomicU32>) -> Self {
+        let sample_rate = inner.sample_rate();
+        let channels = inner.channels();
+        let filters = std::array::from_fn(|band| {
+            let freq = EQ_BANDS_HZ[band].clamp(20.0, (sample_rate as f32 / 2.0) - 100.0);
+            std::array::from_fn(|_| {
+                let coeffs = Coefficients::<f32>::from_params(
+                    FilterType::PeakingEQ(0.0),
+                    (sample_rate as f32).hz(),
+                    freq.hz(),
+                    EQ_Q,
+                ).unwrap_or_else(|_| Coefficients::<f32>::from_params(
+                    FilterType::PeakingEQ(0.0),
+                    (sample_rate as f32).hz(),
+                    1000.0f32.hz(),
+                    EQ_Q,
+                ).unwrap());
+                DirectForm2Transposed::<f32>::new(coeffs)
+            })
+        });
+        Self {
+            inner, sample_rate, channels, gains, enabled, pre_gain,
+            filters,
+            current_gains: [0.0; 10],
+            sample_counter: 0,
+            channel_idx: 0,
+        }
+    }
+
+    fn refresh_if_needed(&mut self) {
+        for band in 0..10 {
+            let gain_db = f32::from_bits(self.gains[band].load(Ordering::Relaxed));
+            if (gain_db - self.current_gains[band]).abs() > 0.01 {
+                self.current_gains[band] = gain_db;
+                let freq = EQ_BANDS_HZ[band].clamp(20.0, (self.sample_rate as f32 / 2.0) - 100.0);
+                if let Ok(coeffs) = Coefficients::<f32>::from_params(
+                    FilterType::PeakingEQ(gain_db),
+                    (self.sample_rate as f32).hz(),
+                    freq.hz(),
+                    EQ_Q,
+                ) {
+                    for ch in 0..2 {
+                        self.filters[band][ch].update_coefficients(coeffs);
+                    }
+                }
+            }
+        }
+    }
+}
+
+impl<S: Source<Item = f32>> Iterator for EqSource<S> {
+    type Item = f32;
+
+    fn next(&mut self) -> Option<f32> {
+        let sample = self.inner.next()?;
+
+        if self.sample_counter % EQ_CHECK_INTERVAL == 0 {
+            self.refresh_if_needed();
+        }
+        self.sample_counter = self.sample_counter.wrapping_add(1);
+
+        if !self.enabled.load(Ordering::Relaxed) {
+            self.channel_idx = (self.channel_idx + 1) % self.channels as usize;
+            return Some(sample);
+        }
+
+        let ch = self.channel_idx.min(1);
+        self.channel_idx = (self.channel_idx + 1) % self.channels as usize;
+
+        let pre_gain_db = f32::from_bits(self.pre_gain.load(Ordering::Relaxed));
+        let pre_gain_factor = 10_f32.powf(pre_gain_db / 20.0);
+        let mut s = sample * pre_gain_factor;
+        for band in 0..10 {
+            s = self.filters[band][ch].run(s);
+        }
+        Some(s.clamp(-1.0, 1.0))
+    }
+}
+
+impl<S: Source<Item = f32>> Source for EqSource<S> {
+    fn current_frame_len(&self) -> Option<usize> { self.inner.current_frame_len() }
+    fn channels(&self) -> u16 { self.channels }
+    fn sample_rate(&self) -> u32 { self.sample_rate }
+    fn total_duration(&self) -> Option<Duration> { self.inner.total_duration() }
+
+    fn try_seek(&mut self, pos: Duration) -> Result<(), rodio::source::SeekError> {
+        // Reset biquad filter state to avoid glitches after seek.
+        for band in 0..10 {
+            let gain_db = f32::from_bits(self.gains[band].load(Ordering::Relaxed));
+            self.current_gains[band] = gain_db;
+            let freq = EQ_BANDS_HZ[band].clamp(20.0, (self.sample_rate as f32 / 2.0) - 100.0);
+            if let Ok(coeffs) = Coefficients::<f32>::from_params(
+                FilterType::PeakingEQ(gain_db),
+                (self.sample_rate as f32).hz(),
+                freq.hz(),
+                EQ_Q,
+            ) {
+                for ch in 0..2 {
+                    self.filters[band][ch] = DirectForm2Transposed::<f32>::new(coeffs);
+                }
+            }
+        }
+        self.channel_idx = 0;
+        self.sample_counter = 0;
+        self.inner.try_seek(pos)
+    }
+}
+
+// ─── DynSource — type-erased Source wrapper ───────────────────────────────────
+//
+// Allows chaining differently-typed sources (with trimming applied) into a
+// single concrete type accepted by EqSource<S: Source<Item=f32>>.
+
+pub(crate) struct DynSource {
+    inner: Box<dyn Source<Item = f32> + Send>,
+    channels: u16,
+    sample_rate: u32,
+}
+
+impl DynSource {
+    pub(crate) fn new(src: impl Source<Item = f32> + Send + 'static) -> Self {
+        let channels = src.channels();
+        let sample_rate = src.sample_rate();
+        Self { inner: Box::new(src), channels, sample_rate }
+    }
+}
+
+impl Iterator for DynSource {
+    type Item = f32;
+    fn next(&mut self) -> Option<f32> { self.inner.next() }
+}
+
+impl Source for DynSource {
+    fn current_frame_len(&self) -> Option<usize> { self.inner.current_frame_len() }
+    fn channels(&self) -> u16 { self.channels }
+    fn sample_rate(&self) -> u32 { self.sample_rate }
+    fn total_duration(&self) -> Option<Duration> { self.inner.total_duration() }
+    fn try_seek(&mut self, pos: Duration) -> Result<(), rodio::source::SeekError> {
+        self.inner.try_seek(pos)
+    }
+}
+
+// ─── EqualPowerFadeIn — per-sample sin(t·π/2) fade-in envelope ───────────────
+//
+// Applied to every new track:
+//   • Crossfade: fade_dur = crossfade_secs  → symmetric equal-power fade-in
+//   • Hard cut:  fade_dur = 5 ms            → micro-fade eliminates DC-click
+//   • Gapless:   fade_dur = 0               → unity gain (no modification)
+//
+// gain(t) = sin(t · π/2),  t ∈ [0, 1)
+// At t = 0 gain = 0, at t = 1 gain = 1.
+// Equal-power property: cos²+sin² = 1 → combined with cos fade-out on Track A
+// the total perceived loudness stays constant across the crossfade.
+
+pub(crate) struct EqualPowerFadeIn<S: Source<Item = f32>> {
+    inner: S,
+    sample_count: u64,
+    fade_samples: u64,
+}
+
+impl<S: Source<Item = f32>> EqualPowerFadeIn<S> {
+    pub(crate) fn new(inner: S, fade_dur: Duration) -> Self {
+        let sample_rate = inner.sample_rate();
+        let channels = inner.channels() as u64;
+        let fade_samples = if fade_dur.is_zero() {
+            0
+        } else {
+            (fade_dur.as_secs_f64() * sample_rate as f64 * channels as f64) as u64
+        };
+        Self { inner, sample_count: 0, fade_samples }
+    }
+}
+
+impl<S: Source<Item = f32>> Iterator for EqualPowerFadeIn<S> {
+    type Item = f32;
+    fn next(&mut self) -> Option<f32> {
+        let sample = self.inner.next()?;
+        let gain = if self.fade_samples == 0 || self.sample_count >= self.fade_samples {
+            1.0
+        } else {
+            let t = self.sample_count as f32 / self.fade_samples as f32;
+            (t * std::f32::consts::FRAC_PI_2).sin()
+        };
+        self.sample_count += 1;
+        Some((sample * gain).clamp(-1.0, 1.0))
+    }
+}
+
+impl<S: Source<Item = f32>> Source for EqualPowerFadeIn<S> {
+    fn current_frame_len(&self) -> Option<usize> { self.inner.current_frame_len() }
+    fn channels(&self) -> u16 { self.inner.channels() }
+    fn sample_rate(&self) -> u32 { self.inner.sample_rate() }
+    fn total_duration(&self) -> Option<Duration> { self.inner.total_duration() }
+    fn try_seek(&mut self, pos: Duration) -> Result<(), rodio::source::SeekError> {
+        // For mid-track seeks: skip straight to unity gain so the new position
+        // plays at full volume immediately — no audible fade-in glitch.
+        // For seeks to the very start (< 100 ms): keep the micro-fade to
+        // suppress any DC-offset click from the fresh decode.
+        if pos.as_millis() < 100 {
+            self.sample_count = 0;
+        } else {
+            self.sample_count = self.fade_samples;
+        }
+        self.inner.try_seek(pos)
+    }
+}
+
+// ─── TriggeredFadeOut — sample-level cos(t·π/2) fade-out triggered externally ─
+//
+// Every track source is wrapped with this. It passes through at unity gain
+// until `trigger` is set to true, at which point it reads `fade_total_samples`
+// and applies a cos(t·π/2) envelope:
+//   gain(t) = cos(t · π/2),  t ∈ [0, 1]
+//   At t = 0 gain = 1, at t = 1 gain = 0.
+// After the fade completes, returns None to exhaust the source.
+//
+// Combined with EqualPowerFadeIn (sin curve) on Track B, this gives a
+// symmetric constant-power crossfade: sin²+cos² = 1.
+
+pub(crate) struct TriggeredFadeOut<S: Source<Item = f32>> {
+    inner: S,
+    trigger: Arc<AtomicBool>,
+    fade_total_samples: Arc<AtomicU64>,
+    fade_progress: u64,
+    fading: bool,
+    cached_total: u64,
+}
+
+impl<S: Source<Item = f32>> TriggeredFadeOut<S> {
+    pub(crate) fn new(inner: S, trigger: Arc<AtomicBool>, fade_total_samples: Arc<AtomicU64>) -> Self {
+        Self {
+            inner,
+            trigger,
+            fade_total_samples,
+            fade_progress: 0,
+            fading: false,
+            cached_total: 0,
+        }
+    }
+}
+
+impl<S: Source<Item = f32>> Iterator for TriggeredFadeOut<S> {
+    type Item = f32;
+    fn next(&mut self) -> Option<f32> {
+        // Check trigger on first fade sample only (avoid atomic load per sample).
+        if !self.fading && self.trigger.load(Ordering::Relaxed) {
+            self.fading = true;
+            self.cached_total = self.fade_total_samples.load(Ordering::Relaxed).max(1);
+            self.fade_progress = 0;
+        }
+
+        if self.fading {
+            if self.fade_progress >= self.cached_total {
+                // Fade complete — exhaust the source.
+                return None;
+            }
+            let sample = self.inner.next()?;
+            let t = self.fade_progress as f32 / self.cached_total as f32;
+            let gain = (t * std::f32::consts::FRAC_PI_2).cos();
+            self.fade_progress += 1;
+            Some((sample * gain).clamp(-1.0, 1.0))
+        } else {
+            self.inner.next()
+        }
+    }
+}
+
+impl<S: Source<Item = f32>> Source for TriggeredFadeOut<S> {
+    fn current_frame_len(&self) -> Option<usize> { self.inner.current_frame_len() }
+    fn channels(&self) -> u16 { self.inner.channels() }
+    fn sample_rate(&self) -> u32 { self.inner.sample_rate() }
+    fn total_duration(&self) -> Option<Duration> { self.inner.total_duration() }
+    fn try_seek(&mut self, pos: Duration) -> Result<(), rodio::source::SeekError> {
+        // If we seek back during a fade, cancel the fade.
+        if self.fading {
+            self.fading = false;
+            self.trigger.store(false, Ordering::Relaxed);
+        }
+        self.fade_progress = 0;
+        self.inner.try_seek(pos)
+    }
+}
+
+// ─── NotifyingSource — sets a flag when the inner iterator is exhausted ───────
+//
+// This is the key mechanism for gapless: the progress task polls `done` to know
+// exactly when source N has finished inside the Sink, without relying on
+// wall-clock estimation or the unreliable `Sink::empty()`.
+
+pub(crate) struct NotifyingSource<S: Source<Item = f32>> {
+    inner: S,
+    done: Arc<AtomicBool>,
+    signalled: bool,
+}
+
+impl<S: Source<Item = f32>> NotifyingSource<S> {
+    pub(crate) fn new(inner: S, done: Arc<AtomicBool>) -> Self {
+        Self { inner, done, signalled: false }
+    }
+}
+
+impl<S: Source<Item = f32>> Iterator for NotifyingSource<S> {
+    type Item = f32;
+    fn next(&mut self) -> Option<f32> {
+        let sample = self.inner.next();
+        if sample.is_none() && !self.signalled {
+            self.signalled = true;
+            self.done.store(true, Ordering::SeqCst);
+        }
+        sample
+    }
+}
+
+impl<S: Source<Item = f32>> Source for NotifyingSource<S> {
+    fn current_frame_len(&self) -> Option<usize> { self.inner.current_frame_len() }
+    fn channels(&self) -> u16 { self.inner.channels() }
+    fn sample_rate(&self) -> u32 { self.inner.sample_rate() }
+    fn total_duration(&self) -> Option<Duration> { self.inner.total_duration() }
+    fn try_seek(&mut self, pos: Duration) -> Result<(), rodio::source::SeekError> {
+        // If we seek backwards the source is no longer exhausted.
+        self.signalled = false;
+        self.done.store(false, Ordering::SeqCst);
+        self.inner.try_seek(pos)
+    }
+}
+
+// ─── CountingSource — atomic sample counter for drift-free position tracking ─
+//
+// Wraps the outermost source and increments a shared AtomicU64 on every sample.
+// The progress task reads this counter and divides by (sample_rate * channels)
+// to get the exact playback position — no wall-clock drift.
+
+pub(crate) struct CountingSource<S: Source<Item = f32>> {
+    inner: S,
+    counter: Arc<AtomicU64>,
+}
+
+impl<S: Source<Item = f32>> CountingSource<S> {
+    pub(crate) fn new(inner: S, counter: Arc<AtomicU64>) -> Self {
+        Self { inner, counter }
+    }
+}
+
+impl<S: Source<Item = f32>> Iterator for CountingSource<S> {
+    type Item = f32;
+    fn next(&mut self) -> Option<f32> {
+        let sample = self.inner.next();
+        if sample.is_some() {
+            self.counter.fetch_add(1, Ordering::Relaxed);
+        }
+        sample
+    }
+}
+
+impl<S: Source<Item = f32>> Source for CountingSource<S> {
+    fn current_frame_len(&self) -> Option<usize> { self.inner.current_frame_len() }
+    fn channels(&self) -> u16 { self.inner.channels() }
+    fn sample_rate(&self) -> u32 { self.inner.sample_rate() }
+    fn total_duration(&self) -> Option<Duration> { self.inner.total_duration() }
+    fn try_seek(&mut self, pos: Duration) -> Result<(), rodio::source::SeekError> {
+        // Reset counter only after confirming the inner seek succeeded.
+        // If we reset first and the seek fails, the counter ends up at the
+        // new position while the decoder is still at the old one — causing
+        // a permanent desync between displayed time and actual audio.
+        let result = self.inner.try_seek(pos);
+        if result.is_ok() {
+            let samples = (pos.as_secs_f64() * self.inner.sample_rate() as f64
+                * self.inner.channels() as f64) as u64;
+            self.counter.store(samples, Ordering::Relaxed);
+        }
+        result
+    }
+}

src-tauri/src/audio/state.rs

@@ -0,0 +1,26 @@
+//! Small shared structs for preload / gapless chain metadata.
+use std::sync::atomic::{AtomicBool, AtomicU64};
+use std::sync::Arc;
+
+pub(crate) struct PreloadedTrack {
+    pub(crate) url: String,
+    pub(crate) data: Vec<u8>,
+}
+
+/// Info about the track that has been appended (chained) to the current Sink
+/// but whose source has not yet started playing (gapless mode only).
+pub(crate) struct ChainedInfo {
+    /// The URL that was chained — used by audio_play to detect a pre-chain hit.
+    pub(crate) url: String,
+    /// Raw file bytes (shared with the chained decoder). Lets manual skip reuse
+    /// them instead of re-downloading after dropping the Sink queue.
+    pub(crate) raw_bytes: Arc<Vec<u8>>,
+    pub(crate) duration_secs: f64,
+    pub(crate) replay_gain_linear: f32,
+    pub(crate) base_volume: f32,
+    /// Set by NotifyingSource when this chained track's source is exhausted.
+    pub(crate) source_done: Arc<AtomicBool>,
+    /// Atomic sample counter for this chained source (swapped into
+    /// samples_played on transition).
+    pub(crate) sample_counter: Arc<AtomicU64>,
+}

src-tauri/src/audio/stream.rs

@@ -0,0 +1,946 @@
+//! Internet radio, ranged/track HTTP readers, ring-buffer producers, and ICY parsing.
+use std::io::{Read, Seek, SeekFrom};
+use std::sync::atomic::{AtomicBool, AtomicU32, AtomicU64, AtomicUsize, Ordering};
+use std::sync::{Arc, Mutex};
+use std::time::{Duration, Instant};
+
+use futures_util::StreamExt;
+use ringbuf::{HeapConsumer, HeapProducer};
+use symphonia::core::io::MediaSource;
+use tauri::{AppHandle, Emitter};
+
+use super::state::PreloadedTrack;
+/// Clears [`AudioEngine::ranged_loudness_seed_hold`] only if it still matches this play.
+struct RangedLoudnessSeedHoldClear {
+    slot: Arc<Mutex<Option<(String, u64)>>>,
+    tid: String,
+    gen: u64,
+}
+
+impl Drop for RangedLoudnessSeedHoldClear {
+    fn drop(&mut self) {
+        if let Ok(mut g) = self.slot.lock() {
+            if matches!(&*g, Some((t, gen)) if t == &self.tid && *gen == self.gen) {
+                *g = None;
+            }
+        }
+    }
+}
+/// 256 KB on the heap — ≈16 s at 128 kbps, ≈6 s at 320 kbps.
+/// Small enough that stale audio drains within a few seconds on reconnect;
+/// large enough to absorb brief network hiccups without stuttering.
+pub(crate) const RADIO_BUF_CAPACITY: usize = 256 * 1024;
+/// Minimum ring buffer for on-demand track streaming starts.
+pub(crate) const TRACK_STREAM_MIN_BUF_CAPACITY: usize = 1024 * 1024;
+/// Cap ring buffer growth when content-length is known.
+pub(crate) const TRACK_STREAM_MAX_BUF_CAPACITY: usize = 32 * 1024 * 1024;
+/// Max bytes kept in memory to promote a completed streamed track for fast replay/seek recovery.
+pub(crate) const TRACK_STREAM_PROMOTE_MAX_BYTES: usize = 64 * 1024 * 1024;
+/// Hot/offline `psysonic-local://` files are read from disk for waveform/LUFS seeding — not the
+/// same heap pressure as retaining a full HTTP capture. FLAC/DSD tracks often exceed 64 MiB;
+/// using the stream-promote cap here skipped analysis entirely (empty seekbar).
+pub(crate) const LOCAL_FILE_PLAYBACK_SEED_MAX_BYTES: usize = 512 * 1024 * 1024;
+/// Consecutive body-stream failures tolerated for track streaming before abort.
+pub(crate) const TRACK_STREAM_MAX_RECONNECTS: u32 = 3;
+/// Seconds at stall threshold while paused before hard-disconnect.
+pub(crate) const RADIO_HARD_PAUSE_SECS: u64 = 5;
+/// AudioStreamReader timeout: if no audio bytes arrive for this long → EOF.
+pub(crate) const RADIO_READ_TIMEOUT_SECS: u64 = 15;
+/// Sleep interval when ring buffer is empty (prevents CPU spin).
+pub(crate) const RADIO_YIELD_MS: u64 = 2;
+
+// ── ICY Metadata State Machine ────────────────────────────────────────────────
+//
+// Shoutcast/Icecast embed metadata every `metaint` audio bytes:
+//
+//   ┌──────────────────────┬───┬─────────────┐
+//   │  audio × metaint     │ N │ meta × N×16 │  (repeating)
+//   └──────────────────────┴───┴─────────────┘
+//
+// N = 0 → no metadata this block.  Metadata bytes are stripped so only
+// pure audio reaches the ring buffer and Symphonia never sees text bytes.
+
+pub(crate) enum IcyState {
+    /// Forwarding audio bytes; `remaining` counts down to the next boundary.
+    ReadingAudio { remaining: usize },
+    /// Next byte is the metadata length multiplier N.
+    ReadingLengthByte,
+    /// Accumulating N×16 metadata bytes.
+    ReadingMetadata { remaining: usize, buf: Vec<u8> },
+}
+
+pub(crate) struct IcyInterceptor {
+    state: IcyState,
+    metaint: usize,
+}
+
+impl IcyInterceptor {
+    fn new(metaint: usize) -> Self {
+        Self { metaint, state: IcyState::ReadingAudio { remaining: metaint } }
+    }
+
+    /// Feed a raw HTTP chunk.
+    /// Appends only audio bytes to `audio_out`.
+    /// Returns `Some(IcyMeta)` when a StreamTitle is extracted.
+    fn process(&mut self, input: &[u8], audio_out: &mut Vec<u8>) -> Option<IcyMeta> {
+        let mut extracted: Option<IcyMeta> = None;
+        let mut i = 0;
+        while i < input.len() {
+            match &mut self.state {
+                IcyState::ReadingAudio { remaining } => {
+                    let n = (input.len() - i).min(*remaining);
+                    audio_out.extend_from_slice(&input[i..i + n]);
+                    i += n;
+                    *remaining -= n;
+                    if *remaining == 0 {
+                        self.state = IcyState::ReadingLengthByte;
+                    }
+                }
+                IcyState::ReadingLengthByte => {
+                    let len_n = input[i] as usize;
+                    i += 1;
+                    self.state = if len_n == 0 {
+                        IcyState::ReadingAudio { remaining: self.metaint }
+                    } else {
+                        IcyState::ReadingMetadata {
+                            remaining: len_n * 16,
+                            buf: Vec::with_capacity(len_n * 16),
+                        }
+                    };
+                }
+                IcyState::ReadingMetadata { remaining, buf } => {
+                    let n = (input.len() - i).min(*remaining);
+                    buf.extend_from_slice(&input[i..i + n]);
+                    i += n;
+                    *remaining -= n;
+                    if *remaining == 0 {
+                        let bytes = std::mem::take(buf);
+                        extracted = parse_icy_meta(&bytes);
+                        self.state = IcyState::ReadingAudio { remaining: self.metaint };
+                    }
+                }
+            }
+        }
+        extracted
+    }
+}
+
+/// ICY metadata parsed from a raw metadata block.
+#[derive(serde::Serialize, Clone)]
+pub(crate) struct IcyMeta {
+    pub title: String,
+    /// `true` when `StreamUrl='0'` — indicates a CDN-injected ad/promo.
+    pub is_ad: bool,
+}
+
+/// Extract `StreamTitle` and `StreamUrl` from a raw ICY metadata block.
+/// Tolerates null padding and non-UTF-8 bytes (lossy conversion).
+fn parse_icy_meta(raw: &[u8]) -> Option<IcyMeta> {
+    let s = String::from_utf8_lossy(raw);
+    let s = s.trim_end_matches('\0');
+
+    const TITLE_TAG: &str = "StreamTitle='";
+    let title_start = s.find(TITLE_TAG)? + TITLE_TAG.len();
+    let title_rest = &s[title_start..];
+    // find (not rfind) — rfind would skip past StreamUrl and corrupt the title
+    let title_end = title_rest.find("';")?;
+    let title = title_rest[..title_end].trim().to_string();
+    if title.is_empty() {
+        return None;
+    }
+
+    const URL_TAG: &str = "StreamUrl='";
+    let stream_url = s.find(URL_TAG).map(|pos| {
+        let rest = &s[pos + URL_TAG.len()..];
+        let end = rest.find("';").unwrap_or(rest.len());
+        rest[..end].trim().to_string()
+    }).unwrap_or_default();
+
+    Some(IcyMeta { title, is_ad: stream_url == "0" })
+}
+
+// ── AudioStreamReader — SPSC consumer → std::io::Read ────────────────────────
+//
+// Bridges HeapConsumer<u8> (non-blocking) into the synchronous Read interface
+// that Symphonia requires.  Designed to run inside tokio::task::spawn_blocking.
+//
+// Empty buffer:  sleeps RADIO_YIELD_MS ms, retries. Never busy-spins.
+// Timeout:       after RADIO_READ_TIMEOUT_SECS with no data → TimedOut.
+// Generation:    if gen_arc != self.gen → Ok(0) (EOF; new track started).
+// Reconnect:     audio_resume sends a fresh HeapConsumer via new_cons_rx.
+//                On the next read() we drain the channel (keep latest) and swap.
+
+pub(crate) struct AudioStreamReader {
+    pub(crate) cons: HeapConsumer<u8>,
+    /// Delivers fresh consumers on hard-pause reconnect (unbounded; drain to latest).
+    /// Wrapped in Mutex so AudioStreamReader is Sync (required by symphonia::MediaSource).
+    /// No real contention: only the audio thread ever calls read().
+    pub(crate) new_cons_rx: Mutex<std::sync::mpsc::Receiver<HeapConsumer<u8>>>,
+    pub(crate) deadline: std::time::Instant,
+    pub(crate) gen_arc: Arc<AtomicU64>,
+    pub(crate) gen: u64,
+    /// Diagnostic tag for logs ("radio" or "track-stream").
+    pub(crate) source_tag: &'static str,
+    /// Optional completion marker: when true and the ring buffer is empty,
+    /// return EOF immediately (used by one-shot track streaming).
+    pub(crate) eof_when_empty: Option<Arc<AtomicBool>>,
+    /// Monotonic byte offset for SeekFrom::Current(0) "tell" (Symphonia probe).
+    pub(crate) pos: u64,
+}
+
+impl Read for AudioStreamReader {
+    fn read(&mut self, buf: &mut [u8]) -> std::io::Result<usize> {
+        // EOF guard: new track started.
+        if self.gen_arc.load(Ordering::SeqCst) != self.gen {
+            return Ok(0);
+        }
+        // Drain reconnect channel; keep only the most recently delivered consumer
+        // so a double-tap of resume doesn't leave stale data in place.
+        let mut newest: Option<HeapConsumer<u8>> = None;
+        while let Ok(c) = self.new_cons_rx.lock().unwrap().try_recv() {
+            newest = Some(c);
+        }
+        if let Some(c) = newest {
+            self.cons = c;
+            self.deadline =
+                std::time::Instant::now() + Duration::from_secs(RADIO_READ_TIMEOUT_SECS);
+        }
+        loop {
+            if self.gen_arc.load(Ordering::SeqCst) != self.gen {
+                return Ok(0);
+            }
+            let available = self.cons.len();
+            if available > 0 {
+                let n = buf.len().min(available);
+                let read = self.cons.pop_slice(&mut buf[..n]);
+                self.pos += read as u64;
+                // Reset deadline: data arrived, so connection is alive.
+                self.deadline =
+                    std::time::Instant::now() + Duration::from_secs(RADIO_READ_TIMEOUT_SECS);
+                return Ok(read);
+            }
+            if self
+                .eof_when_empty
+                .as_ref()
+                .is_some_and(|done| done.load(Ordering::SeqCst))
+            {
+                return Ok(0);
+            }
+            if std::time::Instant::now() >= self.deadline {
+                crate::app_eprintln!(
+                    "[{}] AudioStreamReader: {}s without data → EOF",
+                    self.source_tag,
+                    RADIO_READ_TIMEOUT_SECS
+                );
+                return Err(std::io::Error::new(
+                    std::io::ErrorKind::TimedOut,
+                    format!("{}: no data received", self.source_tag),
+                ));
+            }
+            std::thread::sleep(Duration::from_millis(RADIO_YIELD_MS));
+        }
+    }
+}
+
+impl Seek for AudioStreamReader {
+    fn seek(&mut self, pos: SeekFrom) -> std::io::Result<u64> {
+        match pos {
+            SeekFrom::Current(0) => Ok(self.pos),
+            _ => Err(std::io::Error::new(
+                std::io::ErrorKind::Unsupported,
+                format!("{} stream is not seekable", self.source_tag),
+            )),
+        }
+    }
+}
+
+impl MediaSource for AudioStreamReader {
+    fn is_seekable(&self) -> bool { false }
+    fn byte_len(&self) -> Option<u64> { None }
+}
+
+// ── RangedHttpSource — seekable HTTP-backed MediaSource ──────────────────────
+//
+// Pre-allocates a Vec<u8> of total track size. A background task fills it
+// linearly from offset 0 via streaming HTTP. Read blocks (with timeout) until
+// requested bytes are downloaded; Seek only updates the cursor.
+//
+// Reports is_seekable=true so Symphonia performs time-based seeks via the
+// format reader. Backward seeks: instant (data in buffer). Forward seeks
+// beyond downloaded_to: Read blocks until the linear download catches up.
+//
+// Requires server to have responded with both Content-Length and
+// `Accept-Ranges: bytes` so reconnects can resume via HTTP Range.
+
+pub(crate) struct RangedHttpSource {
+    /// Pre-allocated buffer of total size. Filled linearly from offset 0.
+    pub(crate) buf: Arc<Mutex<Vec<u8>>>,
+    /// Bytes contiguously downloaded from offset 0.
+    pub(crate) downloaded_to: Arc<AtomicUsize>,
+    pub(crate) total_size: u64,
+    pub(crate) pos: u64,
+    /// Set when the download task terminates (success or hard error).
+    pub(crate) done: Arc<AtomicBool>,
+    pub(crate) gen_arc: Arc<AtomicU64>,
+    pub(crate) gen: u64,
+}
+
+impl Read for RangedHttpSource {
+    fn read(&mut self, buf: &mut [u8]) -> std::io::Result<usize> {
+        if self.gen_arc.load(Ordering::SeqCst) != self.gen {
+            return Ok(0);
+        }
+        if self.pos >= self.total_size {
+            return Ok(0);
+        }
+        let max_read = ((self.total_size - self.pos) as usize).min(buf.len());
+        if max_read == 0 {
+            return Ok(0);
+        }
+        let target_end = self.pos + max_read as u64;
+
+        let deadline = Instant::now() + Duration::from_secs(RADIO_READ_TIMEOUT_SECS);
+        loop {
+            if self.gen_arc.load(Ordering::SeqCst) != self.gen {
+                return Ok(0);
+            }
+            let dl = self.downloaded_to.load(Ordering::SeqCst) as u64;
+            if dl >= target_end {
+                break;
+            }
+            // Download finished but our cursor is past downloaded_to (e.g. seek
+            // beyond a partial download that aborted). Return what we have.
+            if self.done.load(Ordering::SeqCst) {
+                if dl > self.pos {
+                    let avail = (dl - self.pos) as usize;
+                    let src = self.buf.lock().unwrap();
+                    let start = self.pos as usize;
+                    buf[..avail].copy_from_slice(&src[start..start + avail]);
+                    drop(src);
+                    self.pos += avail as u64;
+                    return Ok(avail);
+                }
+                return Ok(0);
+            }
+            if Instant::now() >= deadline {
+                return Err(std::io::Error::new(
+                    std::io::ErrorKind::TimedOut,
+                    "ranged-http: no data within timeout",
+                ));
+            }
+            std::thread::sleep(Duration::from_millis(RADIO_YIELD_MS));
+        }
+
+        let src = self.buf.lock().unwrap();
+        let start = self.pos as usize;
+        let end = start + max_read;
+        buf[..max_read].copy_from_slice(&src[start..end]);
+        drop(src);
+        self.pos += max_read as u64;
+        Ok(max_read)
+    }
+}
+
+impl Seek for RangedHttpSource {
+    fn seek(&mut self, pos: SeekFrom) -> std::io::Result<u64> {
+        let new_pos: i64 = match pos {
+            SeekFrom::Start(p) => p as i64,
+            SeekFrom::Current(p) => self.pos as i64 + p,
+            SeekFrom::End(p) => self.total_size as i64 + p,
+        };
+        if new_pos < 0 {
+            return Err(std::io::Error::new(
+                std::io::ErrorKind::InvalidInput,
+                "ranged-http: seek before start",
+            ));
+        }
+        self.pos = (new_pos as u64).min(self.total_size);
+        Ok(self.pos)
+    }
+}
+
+impl MediaSource for RangedHttpSource {
+    fn is_seekable(&self) -> bool { true }
+    fn byte_len(&self) -> Option<u64> { Some(self.total_size) }
+}
+
+// ── LocalFileSource — seekable file-backed MediaSource ───────────────────────
+//
+// Wraps `std::fs::File` so the decoder reads on-demand from disk instead of
+// pre-loading the whole file into a Vec. Used for `psysonic-local://` URLs
+// (offline library + hot playback cache hits) — gives instant track-start
+// because Symphonia only needs to read ~64 KB during probe before playback
+// can begin, vs the previous behaviour of `tokio::fs::read` which blocked
+// until the entire file (often 100+ MB for hi-res FLAC) was in RAM.
+
+pub(crate) struct LocalFileSource {
+    pub(crate) file: std::fs::File,
+    pub(crate) len: u64,
+}
+
+impl Read for LocalFileSource {
+    fn read(&mut self, buf: &mut [u8]) -> std::io::Result<usize> {
+        self.file.read(buf)
+    }
+}
+
+impl Seek for LocalFileSource {
+    fn seek(&mut self, pos: SeekFrom) -> std::io::Result<u64> {
+        self.file.seek(pos)
+    }
+}
+
+impl MediaSource for LocalFileSource {
+    fn is_seekable(&self) -> bool { true }
+    fn byte_len(&self) -> Option<u64> { Some(self.len) }
+}
+
+// ── Pause / Reconnect Coordination ───────────────────────────────────────────
+
+pub(crate) struct RadioSharedFlags {
+    /// Set by audio_pause; cleared by audio_resume.
+    pub(crate) is_paused: AtomicBool,
+    /// Set by download task on hard disconnect; cleared on resume-reconnect.
+    pub(crate) is_hard_paused: AtomicBool,
+    /// Delivers a fresh HeapConsumer<u8> to AudioStreamReader on reconnect.
+    pub(crate) new_cons_tx: Mutex<std::sync::mpsc::Sender<HeapConsumer<u8>>>,
+}
+
+/// Live state for the current radio session, stored in AudioEngine.
+/// Dropping this struct aborts the HTTP download task immediately.
+pub(crate) struct RadioLiveState {
+    pub url: String,
+    pub gen: u64,
+    pub task: tokio::task::JoinHandle<()>,
+    pub flags: Arc<RadioSharedFlags>,
+}
+
+impl Drop for RadioLiveState {
+    fn drop(&mut self) { self.task.abort(); }
+}
+
+// ── HE-AAC / FDK-AAC Fallback ────────────────────────────────────────────────
+//
+// Symphonia 0.5.x: AAC-LC only.  HE-AAC (AAC+) and HE-AACv2 lack SBR/PS →
+// streams play at half speed with muffled audio.
+//
+// With Cargo feature "fdk-aac": FdkAacDecoder is tried first for CODEC_TYPE_AAC.
+// Enable in Cargo.toml:
+//   symphonia-adapter-fdk-aac = { version = "0.1", optional = true }
+//   [features]
+//   fdk-aac = ["dep:symphonia-adapter-fdk-aac"]
+
+// ── Async HTTP Download Task ──────────────────────────────────────────────────
+//
+// Lifecycle:
+//   'outer loop — reconnect on TCP drop (up to MAX_RECONNECTS)
+//   'inner loop — read HTTP chunks → ICY interceptor → push audio to ring buffer
+//
+// Hard-pause detection: if push_slice() returns 0 (buffer full) AND sink is
+// paused AND that condition persists for RADIO_HARD_PAUSE_SECS → disconnect.
+// Sets is_hard_paused = true so audio_resume knows it must reconnect.
+
+pub(crate) async fn radio_download_task(
+    gen: u64,
+    gen_arc: Arc<AtomicU64>,
+    mut initial_response: Option<reqwest::Response>,
+    http_client: reqwest::Client,
+    url: String,
+    mut prod: HeapProducer<u8>,
+    flags: Arc<RadioSharedFlags>,
+    app: AppHandle,
+) {
+    let mut bytes_total: u64 = 0;
+    // Counts consecutive failures (reset on each successful chunk).
+    // laut.fm and similar CDNs force-reconnect every ~700 KB; this is normal.
+    let mut reconnect_count: u32 = 0;
+    const MAX_CONSECUTIVE_FAILURES: u32 = 5;
+    let mut audio_scratch: Vec<u8> = Vec::with_capacity(65_536);
+
+    'outer: loop {
+        if gen_arc.load(Ordering::SeqCst) != gen { return; }
+
+        // ── Obtain response (initial or reconnect) ────────────────────────────
+        let response = match initial_response.take() {
+            Some(r) => r,
+            None => {
+                if reconnect_count >= MAX_CONSECUTIVE_FAILURES {
+                    crate::app_eprintln!("[radio] {MAX_CONSECUTIVE_FAILURES} consecutive failures — giving up");
+                    break 'outer;
+                }
+                tokio::time::sleep(Duration::from_millis(500)).await;
+                if gen_arc.load(Ordering::SeqCst) != gen { return; }
+                match http_client
+                    .get(&url)
+                    .header("Icy-MetaData", "1")
+                    .send()
+                    .await
+                {
+                    Ok(r) if r.status().is_success() => {
+                        crate::app_eprintln!("[radio] reconnected ({bytes_total} B so far)");
+                        r
+                    }
+                    Ok(r) => {
+                        crate::app_eprintln!("[radio] reconnect: HTTP {} — giving up", r.status());
+                        break 'outer;
+                    }
+                    Err(e) => {
+                        crate::app_eprintln!("[radio] reconnect error: {e} — giving up");
+                        break 'outer;
+                    }
+                }
+            }
+        };
+
+        // Parse ICY metaint from each response (consistent across reconnects).
+        let metaint: Option<usize> = response
+            .headers()
+            .get("icy-metaint")
+            .and_then(|v| v.to_str().ok())
+            .and_then(|s| s.parse().ok());
+        let mut icy = metaint.map(IcyInterceptor::new);
+
+        let mut byte_stream = response.bytes_stream();
+        // Stall timer: tracks how long push_slice() returns 0 while paused.
+        let mut stall_since: Option<std::time::Instant> = None;
+
+        'inner: loop {
+            if gen_arc.load(Ordering::SeqCst) != gen { return; }
+
+            // ── Back-pressure + hard-pause detection ──────────────────────────
+            if prod.is_full() {
+                if flags.is_paused.load(Ordering::Relaxed) {
+                    let since = stall_since.get_or_insert(std::time::Instant::now());
+                    if since.elapsed() >= Duration::from_secs(RADIO_HARD_PAUSE_SECS) {
+                        let fill_pct = ((1.0
+                            - prod.free_len() as f32 / RADIO_BUF_CAPACITY as f32)
+                            * 100.0) as u32;
+                        crate::app_eprintln!(
+                            "[radio] hard pause: {fill_pct}% full, \
+                             paused >{RADIO_HARD_PAUSE_SECS}s → disconnecting"
+                        );
+                        flags.is_hard_paused.store(true, Ordering::Release);
+                        return; // Drop HeapProducer → TCP connection released.
+                    }
+                } else {
+                    stall_since = None;
+                }
+                tokio::time::sleep(Duration::from_millis(50)).await;
+                continue 'inner;
+            }
+            stall_since = None;
+
+            // ── Read HTTP chunk ───────────────────────────────────────────────
+            match byte_stream.next().await {
+                Some(Ok(chunk)) => {
+                    bytes_total += chunk.len() as u64;
+                    // Successful data → reset consecutive-failure counter.
+                    reconnect_count = 0;
+                    audio_scratch.clear();
+
+                    if let Some(ref mut interceptor) = icy {
+                        if let Some(meta) = interceptor.process(&chunk, &mut audio_scratch) {
+                            let label = if meta.is_ad { "[Ad]" } else { "" };
+                            crate::app_eprintln!("[radio] ICY StreamTitle: {}{}", label, meta.title);
+                            let _ = app.emit("radio:metadata", &meta);
+                        }
+                    } else {
+                        audio_scratch.extend_from_slice(&chunk);
+                    }
+
+                    // Push with per-chunk back-pressure: yield 5 ms if full mid-chunk.
+                    let mut offset = 0;
+                    while offset < audio_scratch.len() {
+                        if gen_arc.load(Ordering::SeqCst) != gen { return; }
+                        let pushed = prod.push_slice(&audio_scratch[offset..]);
+                        if pushed == 0 {
+                            tokio::time::sleep(Duration::from_millis(5)).await;
+                        } else {
+                            offset += pushed;
+                        }
+                    }
+                }
+                Some(Err(e)) => {
+                    reconnect_count += 1;
+                    crate::app_eprintln!("[radio] stream error: {e} → reconnecting (consecutive #{reconnect_count})");
+                    break 'inner;
+                }
+                None => {
+                    reconnect_count += 1;
+                    crate::app_eprintln!("[radio] stream ended cleanly → reconnecting (consecutive #{reconnect_count})");
+                    break 'inner;
+                }
+            }
+        } // 'inner
+
+        // Do NOT swap the ring buffer here.  The remaining bytes in the buffer
+        // are still valid audio and will drain naturally during reconnect.
+        // Clearing it would cause an immediate underrun/glitch.
+        // The buffer is kept small (RADIO_BUF_CAPACITY) so stale audio drains
+        // within a few seconds rather than minutes.
+    } // 'outer
+
+    crate::app_eprintln!("[radio] download task done ({bytes_total} B total)");
+}
+
+/// One-shot HTTP downloader for track streaming starts.
+///
+/// Pushes response chunks into an SPSC ring buffer consumed by `AudioStreamReader`.
+/// Terminates when:
+/// - generation changes (track superseded),
+/// - response stream ends, or
+/// - response emits an error.
+pub(crate) async fn track_download_task(
+    gen: u64,
+    gen_arc: Arc<AtomicU64>,
+    http_client: reqwest::Client,
+    app: AppHandle,
+    url: String,
+    initial_response: reqwest::Response,
+    mut prod: HeapProducer<u8>,
+    done: Arc<AtomicBool>,
+    promote_cache_slot: Arc<Mutex<Option<PreloadedTrack>>>,
+    normalization_engine: Arc<AtomicU32>,
+    normalization_target_lufs: Arc<AtomicU32>,
+    loudness_pre_analysis_attenuation_db: Arc<AtomicU32>,
+    cache_track_id: Option<String>,
+) {
+    let mut downloaded: u64 = 0;
+    let mut reconnects: u32 = 0;
+    let mut next_response: Option<reqwest::Response> = Some(initial_response);
+    let mut capture: Vec<u8> = Vec::new();
+    let mut capture_over_limit = false;
+    let mut last_partial_loudness_emit = Instant::now() - Duration::from_secs(5);
+    'outer: loop {
+        let response = if let Some(r) = next_response.take() {
+            r
+        } else {
+            let mut req = http_client.get(&url);
+            if downloaded > 0 {
+                req = req.header(reqwest::header::RANGE, format!("bytes={downloaded}-"));
+            }
+            match req.send().await {
+                Ok(r) => r,
+                Err(err) => {
+                    if reconnects >= TRACK_STREAM_MAX_RECONNECTS {
+                        crate::app_eprintln!(
+                            "[audio] streaming reconnect failed after {} attempts: {}",
+                            reconnects, err
+                        );
+                        done.store(true, Ordering::SeqCst);
+                        return;
+                    }
+                    reconnects += 1;
+                    tokio::time::sleep(Duration::from_millis(200)).await;
+                    continue 'outer;
+                }
+            }
+        };
+        if downloaded > 0 && response.status() != reqwest::StatusCode::PARTIAL_CONTENT {
+            crate::app_eprintln!(
+                "[audio] streaming reconnect returned {}, expected 206 for range resume",
+                response.status()
+            );
+            done.store(true, Ordering::SeqCst);
+            return;
+        }
+        if downloaded == 0 && !response.status().is_success() {
+            crate::app_eprintln!("[audio] streaming HTTP {}", response.status());
+            done.store(true, Ordering::SeqCst);
+            return;
+        }
+
+        let mut byte_stream = response.bytes_stream();
+        while let Some(chunk) = byte_stream.next().await {
+            if gen_arc.load(Ordering::SeqCst) != gen {
+                done.store(true, Ordering::SeqCst);
+                return;
+            }
+            let chunk = match chunk {
+                Ok(c) => c,
+                Err(e) => {
+                    if reconnects >= TRACK_STREAM_MAX_RECONNECTS {
+                        crate::app_eprintln!(
+                            "[audio] streaming download error after {} reconnects: {}",
+                            reconnects, e
+                        );
+                        done.store(true, Ordering::SeqCst);
+                        return;
+                    }
+                    reconnects += 1;
+                    crate::app_eprintln!(
+                        "[audio] streaming download error (attempt {}/{}): {} — reconnecting",
+                        reconnects,
+                        TRACK_STREAM_MAX_RECONNECTS,
+                        e
+                    );
+                    next_response = None;
+                    continue 'outer;
+                }
+            };
+            reconnects = 0;
+            let mut offset = 0;
+            while offset < chunk.len() {
+                if gen_arc.load(Ordering::SeqCst) != gen {
+                    done.store(true, Ordering::SeqCst);
+                    return;
+                }
+                let pushed = prod.push_slice(&chunk[offset..]);
+                if pushed == 0 {
+                    tokio::time::sleep(Duration::from_millis(5)).await;
+                } else {
+                    if !capture_over_limit {
+                        if capture.len().saturating_add(pushed) <= TRACK_STREAM_PROMOTE_MAX_BYTES {
+                            let from = offset;
+                            let to = offset + pushed;
+                            capture.extend_from_slice(&chunk[from..to]);
+                        } else {
+                            capture.clear();
+                            capture_over_limit = true;
+                        }
+                    }
+                    if !capture_over_limit
+                        && last_partial_loudness_emit.elapsed() >= Duration::from_millis(crate::audio::helpers::PARTIAL_LOUDNESS_EMIT_INTERVAL_MS)
+                    {
+                        last_partial_loudness_emit = Instant::now();
+                        if normalization_engine.load(Ordering::Relaxed) == 2 {
+                            let target_lufs = f32::from_bits(normalization_target_lufs.load(Ordering::Relaxed));
+                            let pre_db = f32::from_bits(
+                                loudness_pre_analysis_attenuation_db.load(Ordering::Relaxed),
+                            )
+                            .clamp(-24.0, 0.0);
+                            super::helpers::emit_partial_loudness_from_bytes(&app, &url, &capture, target_lufs, pre_db);
+                        }
+                    }
+                    offset += pushed;
+                    downloaded += pushed as u64;
+                }
+            }
+        }
+        if !capture_over_limit && !capture.is_empty() {
+            if let Some(track_id) = cache_track_id {
+                crate::app_deprintln!(
+                    "[stream] legacy stream: capture complete track_id={} capture_mib={:.2} — full-track analysis (cpu-seed queue)",
+                    track_id,
+                    capture.len() as f64 / (1024.0 * 1024.0)
+                );
+                let high = crate::audio::engine::analysis_seed_high_priority_for_track(&app, &track_id);
+                if let Err(e) =
+                    crate::submit_analysis_cpu_seed(app.clone(), track_id.clone(), capture.clone(), high).await
+                {
+                    crate::app_eprintln!("[analysis] track seed failed for {}: {}", track_id, e);
+                }
+            }
+            *promote_cache_slot.lock().unwrap() = Some(PreloadedTrack {
+                url: url.clone(),
+                data: capture,
+            });
+        }
+        done.store(true, Ordering::SeqCst);
+        return;
+    }
+}
+
+/// Linear downloader for `RangedHttpSource`: fills the pre-allocated buffer
+/// from offset 0 to total_size. Reconnects via HTTP Range from the current
+/// `downloaded` offset on transient errors. On completion (full track) the
+/// data is promoted to `stream_completed_cache` for fast replay.
+pub(crate) async fn ranged_download_task(
+    gen: u64,
+    gen_arc: Arc<AtomicU64>,
+    http_client: reqwest::Client,
+    app: AppHandle,
+    _duration_hint: f64,
+    url: String,
+    initial_response: reqwest::Response,
+    buf: Arc<Mutex<Vec<u8>>>,
+    downloaded_to: Arc<AtomicUsize>,
+    done: Arc<AtomicBool>,
+    promote_cache_slot: Arc<Mutex<Option<PreloadedTrack>>>,
+    normalization_engine: Arc<AtomicU32>,
+    normalization_target_lufs: Arc<AtomicU32>,
+    loudness_pre_analysis_attenuation_db: Arc<AtomicU32>,
+    cache_track_id: Option<String>,
+    // When `Some`, ranged playback seeds on completion — defer HTTP backfill for that
+    // track; `None` for large files where ranged skips seed (needs backfill).
+    loudness_seed_hold: Option<Arc<Mutex<Option<(String, u64)>>>>,
+) {
+    let _ranged_loudness_hold_clear = match (loudness_seed_hold.as_ref(), cache_track_id.as_ref()) {
+        (Some(slot), Some(tid)) => {
+            let t = tid.clone();
+            {
+                let mut g = slot.lock().unwrap();
+                *g = Some((t.clone(), gen));
+            }
+            Some(RangedLoudnessSeedHoldClear {
+                slot: Arc::clone(slot),
+                tid: t,
+                gen,
+            })
+        }
+        _ => None,
+    };
+    let total_size = buf.lock().unwrap().len();
+    let mut downloaded: usize = 0;
+    let mut reconnects: u32 = 0;
+    let mut next_response: Option<reqwest::Response> = Some(initial_response);
+    let dl_started = Instant::now();
+    let mut next_progress_mb: usize = 1;
+    let mut last_partial_loudness_emit = Instant::now() - Duration::from_secs(5);
+
+    'outer: loop {
+        let response = if let Some(r) = next_response.take() {
+            r
+        } else {
+            let mut req = http_client.get(&url);
+            if downloaded > 0 {
+                req = req.header(reqwest::header::RANGE, format!("bytes={downloaded}-"));
+            }
+            match req.send().await {
+                Ok(r) => r,
+                Err(err) => {
+                    if reconnects >= TRACK_STREAM_MAX_RECONNECTS {
+                        crate::app_eprintln!(
+                            "[audio] ranged reconnect failed after {} attempts: {}",
+                            reconnects, err
+                        );
+                        break 'outer;
+                    }
+                    reconnects += 1;
+                    tokio::time::sleep(Duration::from_millis(200)).await;
+                    continue 'outer;
+                }
+            }
+        };
+        if downloaded > 0 && response.status() != reqwest::StatusCode::PARTIAL_CONTENT {
+            crate::app_eprintln!(
+                "[audio] ranged reconnect returned {}, expected 206",
+                response.status()
+            );
+            break 'outer;
+        }
+        if downloaded == 0 && !response.status().is_success() {
+            crate::app_eprintln!("[audio] ranged HTTP {}", response.status());
+            break 'outer;
+        }
+
+        let mut byte_stream = response.bytes_stream();
+        while let Some(chunk) = byte_stream.next().await {
+            if gen_arc.load(Ordering::SeqCst) != gen {
+                done.store(true, Ordering::SeqCst);
+                return;
+            }
+            let chunk = match chunk {
+                Ok(c) => c,
+                Err(e) => {
+                    if reconnects >= TRACK_STREAM_MAX_RECONNECTS {
+                        crate::app_eprintln!(
+                            "[audio] ranged dl error after {} reconnects: {}",
+                            reconnects, e
+                        );
+                        break 'outer;
+                    }
+                    reconnects += 1;
+                    crate::app_eprintln!(
+                        "[audio] ranged dl error (attempt {}/{}): {} — reconnecting",
+                        reconnects, TRACK_STREAM_MAX_RECONNECTS, e
+                    );
+                    next_response = None;
+                    continue 'outer;
+                }
+            };
+            reconnects = 0;
+            let writable = total_size.saturating_sub(downloaded);
+            if writable == 0 {
+                break;
+            }
+            let n = chunk.len().min(writable);
+            {
+                let mut b = buf.lock().unwrap();
+                b[downloaded..downloaded + n].copy_from_slice(&chunk[..n]);
+            }
+            downloaded += n;
+            downloaded_to.store(downloaded, Ordering::SeqCst);
+            if downloaded >= crate::audio::helpers::PARTIAL_LOUDNESS_MIN_BYTES
+                && total_size > 0
+                && last_partial_loudness_emit.elapsed() >= Duration::from_millis(crate::audio::helpers::PARTIAL_LOUDNESS_EMIT_INTERVAL_MS)
+            {
+                last_partial_loudness_emit = Instant::now();
+                if normalization_engine.load(Ordering::Relaxed) == 2 {
+                    let target_lufs = f32::from_bits(normalization_target_lufs.load(Ordering::Relaxed));
+                    let start_db = f32::from_bits(loudness_pre_analysis_attenuation_db.load(Ordering::Relaxed))
+                        .clamp(-24.0, 0.0);
+                    if let Some(provisional_db) =
+                        super::helpers::provisional_loudness_gain_from_progress(downloaded, total_size, target_lufs, start_db)
+                    {
+                        let track_key = crate::audio::helpers::playback_identity(&url).unwrap_or_else(|| url.clone());
+                        if crate::audio::ipc::partial_loudness_should_emit(&track_key, provisional_db) {
+                            let _ = app.emit(
+                                "analysis:loudness-partial",
+                                crate::audio::ipc::PartialLoudnessPayload {
+                                    track_id: crate::audio::helpers::playback_identity(&url),
+                                    gain_db: provisional_db,
+                                    target_lufs,
+                                    is_partial: true,
+                                },
+                            );
+                        }
+                    }
+                }
+            }
+            let mb = downloaded / (1024 * 1024);
+            if mb >= next_progress_mb {
+                let pct = (downloaded as f64 / total_size as f64 * 100.0) as u32;
+                crate::app_deprintln!(
+                    "[stream] dl progress: {} MB / {} MB ({}%)",
+                    mb,
+                    total_size / (1024 * 1024),
+                    pct
+                );
+                next_progress_mb = mb + 1;
+            }
+            if downloaded >= total_size {
+                break;
+            }
+        }
+        // Stream ended cleanly (or hit total_size).
+        break 'outer;
+    }
+
+    done.store(true, Ordering::SeqCst);
+
+    crate::app_deprintln!(
+        "[stream] dl done: {} / {} bytes in {:.2}s ({} reconnects)",
+        downloaded,
+        total_size,
+        dl_started.elapsed().as_secs_f64(),
+        reconnects
+    );
+
+    if downloaded == total_size && total_size > 0 && total_size <= TRACK_STREAM_PROMOTE_MAX_BYTES {
+        if let Some(ref tid) = cache_track_id {
+            crate::app_deprintln!(
+                "[stream] ranged: HTTP buffer full track_id={} size_mib={:.2} — cloning {} bytes then full-track analysis (cpu-seed queue; this task awaits completion)",
+                tid,
+                total_size as f64 / (1024.0 * 1024.0),
+                total_size
+            );
+        }
+        let t_clone = Instant::now();
+        let data = buf.lock().unwrap().clone();
+        if total_size > 32 * 1024 * 1024 {
+            crate::app_deprintln!(
+                "[stream] ranged: buffer cloned in_ms={}",
+                t_clone.elapsed().as_millis()
+            );
+        }
+        if let Some(track_id) = cache_track_id {
+            let high = crate::audio::engine::analysis_seed_high_priority_for_track(&app, &track_id);
+            if let Err(e) = crate::submit_analysis_cpu_seed(app.clone(), track_id.clone(), data.clone(), high).await {
+                crate::app_eprintln!("[analysis] ranged seed failed for {}: {}", track_id, e);
+            }
+        }
+        *promote_cache_slot.lock().unwrap() = Some(PreloadedTrack { url, data });
+        crate::app_deprintln!("[stream] promoted to stream_completed_cache for replay");
+    }
+}

src-tauri/src/lib.rs

@@ -4844,29 +4844,29 @@ pub fn run() {
             unregister_global_shortcut,
             mpris_set_metadata,
             mpris_set_playback,
-            audio::audio_play,
+            audio::commands::audio_play,
-            audio::audio_pause,
+            audio::commands::audio_pause,
-            audio::audio_resume,
+            audio::commands::audio_resume,
-            audio::audio_stop,
+            audio::commands::audio_stop,
-            audio::audio_seek,
+            audio::commands::audio_seek,
-            audio::audio_set_volume,
+            audio::commands::audio_set_volume,
-            audio::audio_update_replay_gain,
+            audio::commands::audio_update_replay_gain,
-            audio::audio_set_eq,
+            audio::commands::audio_set_eq,
-            audio::autoeq_entries,
+            audio::commands::autoeq_entries,
-            audio::autoeq_fetch_profile,
+            audio::commands::autoeq_fetch_profile,
-            audio::audio_preload,
+            audio::commands::audio_preload,
-            audio::audio_play_radio,
+            audio::commands::audio_play_radio,
-            audio::audio_preview_play,
+            audio::preview::audio_preview_play,
-            audio::audio_preview_stop,
+            audio::preview::audio_preview_stop,
-            audio::audio_preview_stop_silent,
+            audio::preview::audio_preview_stop_silent,
-            audio::audio_set_crossfade,
+            audio::commands::audio_set_crossfade,
-            audio::audio_set_gapless,
+            audio::commands::audio_set_gapless,
-            audio::audio_set_normalization,
+            audio::commands::audio_set_normalization,
-            audio::audio_list_devices,
+            audio::commands::audio_list_devices,
-            audio::audio_canonicalize_selected_device,
+            audio::commands::audio_canonicalize_selected_device,
-            audio::audio_default_output_device_name,
+            audio::commands::audio_default_output_device_name,
-            audio::audio_set_device,
+            audio::commands::audio_set_device,
-            audio::audio_chain_preload,
+            audio::commands::audio_chain_preload,
             discord::discord_update_presence,
             discord::discord_clear_presence,
             lastfm_request,